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RTCPeerConnection.iceConnectionState - Web APIs
the read-only property rtcpeerconnection.iceconnectionstate returns an enum of type rtciceconnectionstate which state of the ice agent associated with the rtcpeerconnection.
... you can detect when this value has changed by watching for the iceconnectionstatechange event.
... syntax var state = rtcpeerconnection.iceconnectionstate; value the current state of the ice agent and its connection.
...And 14 more matches
RTCPeerConnection.connectionState - Web APIs
the read-only connectionstate property of the rtcpeerconnection interface indicates the current state of the peer connection by returning one of the string values specified by the enum rtcpeerconnectionstate.
... when this property's value changes, a connectionstatechange event is sent to the rtcpeerconnection instance.
... syntax var connectionstate = rtcpeerconnection.connectionstate; value the current state of the connection, as a value from the enum rtcpeerconnectionstate.
...And 11 more matches
RTCPeerConnection: iceconnectionstatechange event - Web APIs
an iceconnectionstatechange event is sent to an rtcpeerconnection object each time the ice connection state changes during the negotiation process.
... the new ice connection state is available in the object's iceconnectionstate} property.
... bubbles no cancelable no interface event event handler property oniceconnectionstatechange one common task performed by the iceconnectionstatechange event listener: to trigger ice restart when the state changes to failed.
...And 9 more matches
RTCPeerConnection.onconnectionstatechange - Web APIs
the rtcpeerconnection.onconnectionstatechange property specifies an eventhandler which is called to handle the connectionstatechange event when it occurs on an instance of rtcpeerconnection.
... this happens whenever the aggregate state of the connection changes.
... the aggregate state is a combination of the states of all of the individual network transports being used by the connection.
...And 3 more matches
RTCPeerConnection.oniceconnectionstatechange - Web APIs
the rtcpeerconnection.oniceconnectionstatechange property is an event handler which specifies a function to be called when the iceconnectionstatechange event is fired on an rtcpeerconnection instance.
... this happens when the state of the connection's ice agent, as represented by the iceconnectionstate property, changes.
... syntax rtcpeerconnection.oniceconnectionstatechange = eventhandler; value this event handler can be set to function which is passed a single input parameter: an event object describing the iceconnectionstatechange event which occurred.
...And 3 more matches
RTCPeerConnection: connectionstatechange event - Web APIs
the connectionstatechange event is sent to the ontrack event handler on an rtcpeerconnection object after a new track has been added to an rtcrtpreceiver which is part of the connection.
... the new connection state can be found in connectionstate, and is one of the strings in the rtcpeerconnectionstate enumerated type.
... bubbles no cancelable no interface event event handler onconnectionstatechange examples for an rtcpeerconnection, pc, this example sets up a handler for connectionstatechange messages to handle changes to the connectivity of the webrtc session.
...And 2 more matches
RTCPeerConnection - Web APIs
the rtcpeerconnection interface represents a webrtc connection between the local computer and a remote peer.
... it provides methods to connect to a remote peer, maintain and monitor the connection, and close the connection once it's no longer needed.
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...And 94 more matches
Connection management in HTTP/1.x - HTTP
connection management is a key topic in http: opening and maintaining connections largely impacts the performance of web sites and web applications.
... in http/1.x, there are several models: short-lived connections, persistent connections, and http pipelining.
... http mostly relies on tcp for its transport protocol, providing a connection between the client and the server.
...And 37 more matches
mozIStorageConnection
storage/public/mozistorageconnection.idlscriptable this interface represents a database connection attached to a specific file or an in-memory database.
...removeprogresshandler(); void rollbacktransaction(); void setgrowthincrement(in print32 aincrement, in autf8string adatabasename); mozistorageprogresshandler setprogresshandler(in print32 agranularity, in mozistorageprogresshandler ahandler); boolean tableexists(in autf8string atablename); attributes attribute type description connectionready boolean indicates if the connection is open and ready to use.
... this will be false if the connection failed to open, or it has been closed.
...And 27 more matches
Establishing a connection: The WebRTC perfect negotiation pattern - Web APIs
because webrtc doesn't mandate a specific transport mechanism for signaling during the negotiation of a new peer connection, it's highly flexible.
...this article introduces webrtc perfect negotiation, describing how it works and why it's the recommended way to negotiate a webrtc connection between peers, and provides sample code to demonstrate the technique.
...negotiation is an inherently asymmetric operation: one side needs to serve as the "caller" while the other peer is the "callee." the perfect negotiation pattern smooths this difference away by separating that difference out into independent negotiation logic, so that your application doesn't need to care which end of the connection it is.
...And 20 more matches
RTCPeerConnection.setRemoteDescription() - Web APIs
the rtcpeerconnection method setremotedescription() sets the specified session description as the remote peer's current offer or answer.
... the description specifies the properties of the remote end of the connection, including the media format.
...keep in mind that if setremotedescription() is called while a connection is already in place, it means renegotiation is underway (possibly to adapt to changing network conditions).
...And 16 more matches
RTCPeerConnection.addTrack() - Web APIs
the rtcpeerconnection method addtrack() adds a new media track to the set of tracks which will be transmitted to the other peer.
... note: adding a track to a connection triggers renegotiation by firing a negotiationneeded event.
... syntax rtpsender = rtcpeerconnection.addtrack(track, stream...); parameters track a mediastreamtrack object representing the media track to add to the peer connection.
...And 12 more matches
RTCPeerConnection.signalingState - Web APIs
the read-only signalingstate property on the rtcpeerconnection interface returns one of the string values specified by the rtcsignalingstate enum; these values describe the state of the signaling process on the local end of the connection while connecting or reconnecting to another peer.
...for example, if you receive an answer while the signalingstate isn't "have-local-offer", you know that something is wrong, since you should only receive answers after creating an offer but before an answer has been received and passed into rtcpeerconnection.setlocaldescription().
... in addition, when the value of this property changes, a signalingstatechange event is sent to the rtcpeerconnection instance.
...And 10 more matches
RTCPeerConnection() - Web APIs
the rtcpeerconnection() constructor returns a newly-created rtcpeerconnection, which represents a connection between the local device and a remote peer.
... syntax pc = new rtcpeerconnection([configuration]); parameters configuration optional an rtcconfiguration dictionary providing options to configure the new connection.
... certificates optional an array of objects of type rtccertificate which are used by the connection for authentication.
...And 9 more matches
RTCPeerConnection.setLocalDescription() - Web APIs
the rtcpeerconnection method setlocaldescription() changes the local description associated with the connection.
... this description specifies the properties of the local end of the connection, including the media format.
... if setlocaldescription() is called while a connection is already in place, it means renegotiation is underway (possibly to adapt to changing network conditions).
...And 8 more matches
RTCPeerConnection.createOffer() - Web APIs
the createoffer() method of the rtcpeerconnection interface initiates the creation of an sdp offer for the purpose of starting a new webrtc connection to a remote peer.
... the sdp offer includes information about any mediastreamtracks already attached to the webrtc session, codec, and options supported by the browser, and any candidates already gathered by the ice agent, for the purpose of being sent over the signaling channel to a potential peer to request a connection or to update the configuration of an existing connection.
... syntax apromise = mypeerconnection.createoffer([options]); mypeerconnection.createoffer(successcallback, failurecallback, [options]) parameters options optional an rtcofferoptions dictionary providing options requested for the offer.
...And 7 more matches
RTCPeerConnection.getStats() - Web APIs
the rtcpeerconnection method getstats() returns a promise which resolves with data providing statistics about either the overall connection or about the specified mediastreamtrack.
... syntax promise = rtcpeerconnection.getstats(selector) parameters selector optional a mediastreamtrack for which to gather statistics.
... if this is null (the default value), statistics will be gathered for the entire rtcpeerconnection.
...And 7 more matches
RTCPeerConnection.setConfiguration() - Web APIs
the rtcpeerconnection.setconfiguration() method sets the current configuration of the rtcpeerconnection based on the values included in the specified rtcconfiguration object.
... this lets you change the ice servers used by the connection and which transport policies to use.
...two potential scenarios in which this might be done: the rtcpeerconnection was instantiated without specifying any ice servers.
...And 7 more matches
RTCPeerConnection: icecandidateerror event - Web APIs
the webrtc api event icecandidateerror is sent to an rtcpeerconnection if an error occurs while performing ice negotiations through a stun or turn server.
... the event object is of type rtcpeerconnectioniceerrorevent, and contains information describing the error in some amount of detail.
... bubbles no cancelable no interface rtcpeerconnectioniceerrorevent event handler property rtcpeerconnection.onicecandidateerror description the error object's errorcode property is one of the numeric stun error codes.
...And 6 more matches
Creating Sandboxed HTTP Connections
introduction starting with gecko 1.8.1 (firefox 2), it is possible to create sandboxed http connections which don't affect the user's cookies.
... this article will cover the basics of doing http connections from xpcom javascript, and should easily translate to c++ xpcom.
... setting up an http connection the first step in setting up an http connection from an url (stored in a string) is to create an nsiuri out of it.
...And 5 more matches
RTCPeerConnection.addStream() - Web APIs
the obsolete rtcpeerconnection method addstream() adds a mediastream as a local source of audio or video.
...if the signalingstate is set to stable, the event negotiationneeded is sent on the rtcpeerconnection to indicate that ice negotiation must be repeated to consider the new stream.
... syntax rtcpeerconnection.addstream(mediastream); parameters mediastream a mediastream object indicating the stream to add to the webrtc peer connection.
...And 5 more matches
RTCPeerConnection.getConfiguration() - Web APIs
the rtcpeerconnection.getconfiguration() method returns an rtcconfiguration object which indicates the current configuration of the rtcpeerconnection on which the method is called.
... the returned configuration is the last configuration applied via setconfiguration(), or if setconfiguration() hasn't been called, the configuration the rtcpeerconnection was constructed with.
... the configuration includes a list of the ice servers used by the connection, information about transport policies, and identity information.
...And 5 more matches
RTCPeerConnection.removeTrack() - Web APIs
the rtcpeerconnection.removetrack() method tells the local end of the connection to stop sending media from the specified track, without actually removing the corresponding rtcrtpsender from the list of senders as reported by rtcpeerconnection.getsenders().
... if the track is already stopped, or is not in the connection's senders list, this method has no effect.
... if the connection has already been negotiated (signalingstate is set to "stable"), it is marked as needing to be negotiated again; the remote peer won't experience the change until this negotiation occurs.
...And 5 more matches
RTCPeerConnection.restartIce() - Web APIs
the webrtc api's rtcpeerconnection interface offers the restartice() method to allow a web application to easily request that ice candidate gathering be redone on both ends of the connection.
... restartice() causes the negotiationneeded event to be fired on the rtcpeerconnection to inform the application that it should perform negotiation using its signaling channel.
... if negotiation fails to complete—either due to rollback or because incoming offers are in the process of being negotiated—the rtcpeerconnection will remember that you requested ice restart.
...And 5 more matches
Connection - HTTP
the connection general header controls whether or not the network connection stays open after the current transaction finishes.
... if the value sent is keep-alive, the connection is persistent and not closed, allowing for subsequent requests to the same server to be done.
... connection-specific header fields such as connection and keep-alive are prohibited in http/2.
...And 5 more matches
RTCPeerConnection.onaddstream - Web APIs
the rtcpeerconnection.onaddstream event handler is a property containing the code to execute when the addstream event, of type mediastreamevent, is received by this rtcpeerconnection.
... such an event is sent when a mediastream is added to this connection by the remote peer.
... important: this property has been removed from the specification; you should now use rtcpeerconnection.ontrack to watch for track events instead.
...And 4 more matches
RTCPeerConnection.onnegotiationneeded - Web APIs
the rtcpeerconnection interface's onnegotiationneeded property is an eventlistener which specifies a function which is called to handle the negotiationneeded event when it occurs on an rtcpeerconnection instance.
... most commonly, the negotiationneeded event is fired after a send track is added to the rtcpeerconnection.
... syntax rtcpeerconnection.onnegotiationneeded = eventhandler; value this should be set to a function you provide which is passed a single parameter: an event object containing the negotiationneeded event.
...And 4 more matches
RTCPeerConnectionIceErrorEvent.address - Web APIs
the rtcpeerconnectioniceerrorevent property address is a string which indicates the local ip address being used to communicate with the stun or turn server during negotiations.
... syntax let address = rtcpeerconnectioniceerrorevent.address; value a domstring which specifies the local ip address of the network connection to the ice server with which negotiations were occurring when the error occurred.
... this address identifies the network interface on the local device which is being used to attempt to establish the connection to the remote peer.
...And 4 more matches
PerformanceResourceTiming.secureConnectionStart - Web APIs
the secureconnectionstart read-only property returns a timestamp immediately before the browser starts the handshake process to secure the current connection.
... if a secure connection is not used, the property returns zero.
... syntax resource.secureconnectionstart; return value if the resource is fetched over a secure connection, a domhighrestimestamp immediately before the browser starts the handshake process to secure the current connection.
...And 3 more matches
RTCPeerConnection.addIceCandidate() - Web APIs
when a web site or app using rtcpeerconnection receives a new ice candidate from the remote peer over its signaling channel, it delivers the newly-received candidate to the browser's ice agent by calling rtcpeerconnection.addicecandidate().
... this adds this new remote candidate to the rtcpeerconnection's remote description, which describes the state of the remote end of the connection.
... during negotiation, your app will likely receive many candidates which you'll deliver to the ice agent in this way, allowing it to build up a list of potential connection methods.
...And 3 more matches
RTCPeerConnection.canTrickleIceCandidates - Web APIs
the read-only rtcpeerconnection property cantrickleicecandidates returns a boolean which indicates whether or not the remote peer can accept trickled ice candidates.
... this property is only set after having called rtcpeerconnection.setremotedescription().
... syntax var cantrickle = rtcpeerconnection.cantrickleicecandidates; value a boolean that is true if the remote peer can accept trickled ice candidates and false if it cannot.
...And 3 more matches
RTCPeerConnection.close() - Web APIs
the rtcpeerconnection.close() method closes the current peer connection.
... syntax peerconnection.close(); this method has no parameters, and returns nothing.
... calling this method terminates the rtcpeerconnection's ice agent, ending any ongoing ice processing and any active streams.
...And 3 more matches
RTCPeerConnection.createAnswer() - Web APIs
the createanswer() method on the rtcpeerconnection interface creates an sdp answer to an offer received from a remote peer during the offer/answer negotiation of a webrtc connection.
... syntax apromise = rtcpeerconnection.createanswer([options]); rtcpeerconnection.createanswer(successcallback, failurecallback[, options]); parameters options optional an object which contains options which customize the answer; this is based on the rtcansweroptions dictionary.
... failurecallback an rtcpeerconnectionerrorcallback which will be passed a single domexception object explaining why the request to create an answer failed.
...And 3 more matches
RTCPeerConnection.createDataChannel() - Web APIs
the createdatachannel() method on the rtcpeerconnection interface creates a new channel linked with the remote peer, over which any kind of data may be transmitted.
... if the new data channel is the first one added to the connection, renegotiation is started by delivering a negotiationneeded event.
... syntax datachannel = rtcpeerconnection.createdatachannel(label[, options]); parameters label a human-readable name for the channel.
...And 3 more matches
RTCPeerConnection.getSenders() - Web APIs
the rtcpeerconnection method getsenders() returns an array of rtcrtpsender objects, each of which represents the rtp sender responsible for transmitting one track's data.
... syntax var senders = rtcpeerconnection.getsenders(); return value an array of rtcrtpsender objects, one for each track on the connection.
... the array is empty if there are no rtp senders on the connection.
...And 3 more matches
RTCPeerConnection.getTransceivers() - Web APIs
the rtcpeerconnection interface's gettransceivers() method returns a list of the rtcrtptransceiver objects being used to send and receive data on the connection.
... syntax transceiverlist = rtcpeerconnection.gettransceivers(); parameters none.
... return value an array of the rtcrtptransceiver objects representing the transceivers handling sending and receiving all media on the rtcpeerconnection.
...And 3 more matches
RTCPeerConnection.iceGatheringState - Web APIs
the read-only property rtcpeerconnection.icegatheringstate returns an enum of type rtcicegatheringstate that describes connection's ice gathering state.
... syntax var state = rtcpeerconnection.icegatheringstate; value the possible values are those of an enum of type rtcicegatheringstate.
... rtcicegatheringstate enum the rtcicegatheringstate enum defines string constants which reflect the current status of ice gathering, as returned using the rtcpeerconnection.icegatheringstate property.
...And 3 more matches
RTCPeerConnection: icegatheringstatechange event - Web APIs
the icegatheringstatechange event is sent to the onicegatheringstatechange event handler on an rtcpeerconnection when the state of the ice candidate gathering process changes.
... this signifies that the value of the connection's icegatheringstate property has changed.
... when ice firststarts to gather connection candidates, the value changes from new to gathering to indicate that the process of collecting candidate configurations for the connection has begun.
...And 3 more matches
RTCPeerConnection: icecandidate event - Web APIs
an icecandidate event is sent to an rtcpeerconnection when an rtcicecandidate has been identified and added to the local peer by a call to rtcpeerconnection.setlocaldescription().
... bubbles no cancelable no interface rtcpeerconnectioniceevent event handler property rtcpeerconnection.onicecandidate description there are three reasons why the icecandidate event is fired on an rtcpeerconnection.
... rtcpeerconnection.onicecandidate = (event) => { if (event.candidate) { sendcandidatetoremotepeer(event.candidate) } else { /* there are no more candidates coming during this negotiation */ } } the remote peer, upon receiving the candidate, will add the candidate to its candidate pool by calling addicecandidate(), passing in the candidate string you have passed along using the signaling server.
...And 3 more matches
RTCPeerConnection: negotiationneeded event - Web APIs
a negotiationneeded event is sent to the rtcpeerconnection when negotiation of the connection through the signaling channel is required.
... this occurs both during the initial setup of the connection as well as any time a change to the communication environment requires reconfiguring the connection.
... bubbles no cancelable no interface event event handler property rtcpeerconnection.onnegotiationneeded the negotiationneeded event is first dispatched to the rtcpeerconnection when media is first added to the connection.
...And 3 more matches
RTCPeerConnection.pendingLocalDescription - Web APIs
the read-only property rtcpeerconnection.pendinglocaldescription returns an rtcsessiondescription object describing a pending configuration change for the local end of the connection.
... this does not describe the connection as it currently stands, but as it may exist in the near future.
... use rtcpeerconnection.currentlocaldescription or rtcpeerconnection.localdescription to get the current state of the endpoint.
...And 3 more matches
RTCPeerConnection.pendingRemoteDescription - Web APIs
the read-only property rtcpeerconnection.pendingremotedescription returns an rtcsessiondescription object describing a pending configuration change for the remote end of the connection.
... this does not describe the connection as it currently stands, but as it may exist in the near future.
... use rtcpeerconnection.currentremotedescription or rtcpeerconnection.remotedescription to get the current session description for the remote endpoint.
...And 3 more matches
RTCPeerConnectionIceErrorEvent - Web APIs
the rtcpeerconnectioniceerrorevent interface—based upon the event interface—provides details pertaining to an ice error announced by sending an icecandidateerror event to the rtcpeerconnection object.
... constructor rtcpeerconnectioniceerrorevent() creates and returns a new rtcpeerconnectioniceerrorevent object, with its type and other properties initialized as specified in the parameters.
... properties the rtcpeerconnectioniceerrorevent interface includes the properties found on the event interface, as well as the following properties: address read only a domstring providing the local ip address used to communicate with the stun or turn server being used to negotiate the connection, or null if the local ip address has not yet been exposed as part of a local ice candidate.
...And 3 more matches
RTCPeerConnectionIceEvent - Web APIs
the rtcpeerconnectioniceevent interface represents events that occurs in relation to ice candidates with the target, usually an rtcpeerconnection.
... properties a rtcpeerconnectioniceevent being an event, this event also implements these properties.
... rtcpeerconnectioniceevent.candidate read only contains the rtcicecandidate containing the candidate associated with the event, or null if this event indicates that there are no further candidates to come.
...And 3 more matches
RTCPeerConnection: addstream event - Web APIs
the obsolete addstream event is sent to an rtcpeerconnection when new media, in the form of a mediastream object, has been added to it.
...you should instead watch for the track event, which is sent for each media track added to the rtcpeerconnection.
... bubbles no cancelable no interface mediastreamevent event handler property rtcpeerconnection.onaddstream you can, similarly, watch for streams to be removed from the connection by monitoring the removestream event.
...And 2 more matches
RTCPeerConnection.currentLocalDescription - Web APIs
the read-only property rtcpeerconnection.currentlocaldescription returns an rtcsessiondescription object describing the local end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
... to change the currentlocaldescription, call rtcpeerconnection.setlocaldescription(), which triggers a series of events which leads to this value being set.
... unlike rtcpeerconnection.localdescription, this value represents the actual current state of the local end of the connection; localdescription may specify a description which the connection is currently in the process of switching over to.
...And 2 more matches
RTCPeerConnection.currentRemoteDescription - Web APIs
the read-only property rtcpeerconnection.currentremotedescription returns an rtcsessiondescription object describing the remote end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
... to change the currentremotedescription, call rtcpeerconnection.setremotedescription(), which triggers a series of events which leads to this value being set.
... unlike rtcpeerconnection.remotedescription, this value represents the actual current state of the local end of the connection; remotedescription may specify a description which the connection is currently in the process of switching over to.
...And 2 more matches
RTCPeerConnection.onsignalingstatechange - Web APIs
the onsignalingstatechange event handler property of the rtcpeerconnection interface specifies a function to be called when the signalingstatechange event occurs on an rtcpeerconnection interface.
... the function receives as input the event object of type event; this event is sent when the peer connection's signalingstate changes, which may happen either because of a call to setlocaldescription() or to setremotedescription().
... syntax rtcpeerconnection.onsignalingstatechange = errorhandler; value set this to a function which you provide that receives an event object as input; this contains the signalingstatechange event.
...And 2 more matches
RTCPeerConnection: peeridentity event - Web APIs
the peeridentity event is sent to the connection concerned when peer identity has been set and verified on it.
... the new identiy can be access using the rtcpeerconnection.peeridentity property.
... an event handler for this event can be added via the rtcpeerconnection.onpeeridentity property.
...And 2 more matches
RTCPeerConnectionIceEvent() - Web APIs
the rtcpeerconnectioniceevent() constructor creates a new rtcpeerconnectioniceevent.
... syntax var event = new rtcpeerconnectioniceevent(type, options); parameters type is a domstring containing the name of the event, like "icecandidate".
... options a dictionary of type rtcpeerconnectioninit, which may contain one or more of the following fields: "candidate" (optional, default is null): a rtcicecandidate representing the ice candidate being concerned by the event.
...And 2 more matches
Effective connection type - MDN Web Docs Glossary: Definitions of Web-related terms
effective connection type (ect) refers to the measured network performance, returning a cellular connection type, like 3g, even if the actual connection is tethered broadband or wifi, based on the time between the browser requesting a page and effective type of the connection.
... table of effective connection types ect minimum rtt maximum downlink explanation slow-2g 2000ms 50 kbps the network is suited for small transfers only such as text-only pages.
... effectivetype is a property of the network information api, exposed to javascript via the navigator.connection object.
... to see your effective connection type, open the console of the developer tools of a supporting browser and enter the following: navigator.connection.effectivetype; see also: network information api networkinformation networkinformation.effectivetype ...
Navigator.connection - Web APIs
the navigator.connection read-only property returns a networkinformation object containing information about the system's connection, such as the current bandwidth of the user's device or whether the connection is metered.
... this could be used to select high definition content or low definition content based on the user's connection.
... syntax networkinformation = navigator.connection value a networkinformation object.
... specifications specification status comment network information apithe definition of 'navigator.connection' in that specification.
RTCPeerConnection.addTransceiver() - Web APIs
the rtcpeerconnection method addtransceiver() creates a new rtcrtptransceiver and adds it to the set of transceivers associated with the rtcpeerconnection.
... syntax rtptransceiver = rtcpeerconnection.addtransceiver(trackorkind, init); parameters trackorkind a mediastreamtrack to associate with the transceiver, or a domstring which is used as the kind of the receiver's track, and by extension of the rtcrtpreceiver itself.
... streams optional a list of mediastream objects to add to the transceiver'srtcrtpreceiver; when the remote peer's rtcpeerconnection's track event occurs, these are the streams that will be specified by that event.
... specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.addtransceiver()' in that specification.
RTCPeerConnection.generateCertificate() - Web APIs
the generatecertificate() method of the rtcpeerconnection interface creates and stores an x.509 certificate and corresponding private key then returns an rtccertificate, providing access to it.
... syntax var cert = rtcpeerconnection.generatecertificate(keygenalgorithm) parameters keygenalgorithm a domstring identifying the algorithm to use in creating the key.
... rtcpeerconnection.generatecertificate() is a static method, so it is always called on the rtcpeerconnection interface itself, not an instance thereof.
... example rtcpeerconnection.generatecertificate({ name: 'rsassa-pkcs1-v1_5', hash: 'sha-256', moduluslength: 2048, publicexponent: new uint8array([1, 0, 1]) }).then(function(cert) { var pc = new rtcpeerconnection({certificates: [cert]}); }); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'generatecertificate()' in that specification.
RTCPeerConnection.getIdentityAssertion() - Web APIs
the rtcpeerconnection.getidentityassertion() method initiates the gathering of an identity assertion.
... it is not expected for the application dealing with the rtcpeerconnection: this is automatically done; an explicit call only allows to anticipate the need.
... example var pc = new peerconnection(); pc.getidentityassertion(); // not mandatory, but we know that we will need it in the future.
... specifications specification status comment identity for webrtcthe definition of 'rtcpeerconnection.getidentityassertion()' in that specification.
RTCPeerConnection.getReceivers() - Web APIs
the rtcpeerconnection.getreceivers() method returns an array of rtcrtpreceiver objects, each of which represents one rtp receiver.
... each rtp receiver manages the reception and decoding of data for a mediastreamtrack on an rtcpeerconnection syntax var receivers = rtcpeerconnection.getreceivers(); return value an array of rtcrtpreceiver objects, one for each track on the connection.
... the array is empty if there are no rtp receivers on the connection.
... example tbd specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.getreceivers()' in that specification.
RTCPeerConnection.getStreamById() - Web APIs
the rtcpeerconnection.getstreambyid() method returns the mediastream with the given id that is associated with local or remote end of the connection.
... this property has been replaced with the rtcpeerconnection.getlocalstreams and rtcpeerconnection.getremotestreams properties.
... example var stream = pc.getstreambyid(mytrackid); if (stream) { console.log("found stream: " + stream.id); } polyfill running the following code before any other code will create rtcpeerconnection.prototype.getstreambyid() if it's not natively available.
... // from: https://bugs.chromium.org/p/chromium/issues/detail?id=698163&desc=5#c10 rtcpeerconnection.prototype.getstreambyid = function(id) { try { var localstreams = this.getlocalstreams(); var remotestreams = this.getremotestreams(); var i; for (i = 0; i < localstreams.length; i++) { if (localstreams[i].id == id) return localstreams[i]; } for (i = 0; i < remotestreams.length; i++) { if (remotestreams[i].id == id) return remotestreams[i]; } } catch(e) {} return null; } ...
RTCPeerConnection.ondatachannel - Web APIs
the rtcpeerconnection.ondatachannel property is an eventhandler which specifies a function which is called when the datachannel event occurs on an rtcpeerconnection.
... this event, of type rtcdatachannelevent, is sent when an rtcdatachannel is added to the connection by the remote peer calling createdatachannel().
... syntax rtcpeerconnection.ondatachannel = function; value set this property to be a function you provide which receives as input a single parameter: an rtcdatachannelevent which provides in its channel property the rtcdatachannel which has been created.
... example pc.ondatachannel = function(ev) { console.log('data channel is created!'); ev.channel.onopen = function() { console.log('data channel is open and ready to be used.'); }; }; specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.ondatachannel' in that specification.
RTCPeerConnection.onicecandidate - Web APIs
the rtcpeerconnection property onicecandidate property is an eventhandler which specifies a function to be called when the icecandidate event occurs on an rtcpeerconnection instance.
... syntax rtcpeerconnection.onicecandidate = eventhandler; value this should be set to a function which you provide that accepts as input an rtcpeerconnectioniceevent object representing the icecandidate event.
...when this happens, the connection's icegatheringstate has also changed to complete.
... specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.onicecandidate' in that specification.
RTCPeerConnection.onicegatheringstatechange - Web APIs
the rtcpeerconnection.onicegatheringstatechange property is an eventhandler which specifies a function to be called when the icegatheringstatechange event is sent to an rtcpeerconnection instance.
... syntax rtcpeerconnection.onicegatheringstatechange = eventhandler; value a function you provide which is passed a single parameter: an event object containing the icegatheringstatechange event.
... you can determine the new state of ice gathering by looking at the value of the rtcpeerconnection.icegatheringstate property.
...nown"; switch(pc.icegatheringstate) { case "new": case "complete": label = "idle"; break; case "gathering": label = "determining route"; break; } document.getelementbyid("icestatus").innerhtml = label; } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.onicegatheringstatechange' in that specification.
RTCPeerConnection.onremovestream - Web APIs
the rtcpeerconnection api is now track-based, so having zero tracks in the remote stream is equivalent to the remote stream being removed and the old removestream event.
... the rtcpeerconnection.onremovestream event handler is a property containing the code to execute when the removestream event, of type mediastreamevent, is received by this rtcpeerconnection.
... such an event is sent when a mediastream is removed from this connection.
... syntax peerconnection.onremovestream = function; values function is the name of a user-defined function, without the () suffix or any parameters, or an anonymous function declaration, such as function(event) {...}.
RTCPeerConnection.ontrack - Web APIs
the rtcpeerconnection property ontrack is an eventhandler which specifies a function to be called when the track event occurs, indicating that a track has been added to the rtcpeerconnection.
... the function receives as input the event object, of type rtctrackevent; this event is sent when a new incoming mediastreamtrack has been created and associated with an rtcrtpreceiver object which has been added to the set of receivers on connection.
... syntax rtcpeerconnection.ontrack = eventhandler; value set ontrack to be a function you provide that accepts as input a rtctrackevent object describing the new track and how it's being used.
... specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.ontrack' in that specification.
RTCPeerConnection.peerIdentity - Web APIs
the read-only rtcpeerconnection property peeridentity returns a javascript promise that resolves to an rtcidentityassertion which contains a domstring identifying the remote peer.
... once this promise resolves successfully, the resulting identity is the target peer identity and cannot change for the duration of the connection.
... syntax var identity = rtcpeerconnection.peeridentity; value a javascript promise which resolves to an rtcidentityassertion that describes the remote peer's identity.
... let pc = new rtcpeerconnection(); /* ...
RTCPeerConnection.remoteDescription - Web APIs
the read-only property rtcpeerconnection.remotedescription returns a rtcsessiondescription describing the session (which includes configuration and media information) for the remote end of the connection.
... the returned value typically reflects a remote description which has been received over the signaling server (as either an offer or an answer) and then put into effect by your code calling rtcpeerconnection.setremotedescription() in response.
... syntax var sessiondescription = peerconnection.remotedescription; on a more fundamental level, the returned value is the value of rtcpeerconnection.pendingremotedescription if that property isn't null; otherwise, the value of rtcpeerconnection.currentremotedescription is returned.
... var pc = new rtcpeerconnection(); … var sd = pc.remotedescription; if (sd) { alert("remote session: type='" + sd.type + "'; sdp description='" + sd.sdp + "'"); } else { alert("no remote session yet."); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.remotedescription' in that specification.
RTCPeerConnection.removeStream() - Web APIs
the rtcpeerconnection.removestream() method removes a mediastream as a local source of audio or video.
...if the signalingstate is set to "stable", the event negotiationneeded is sent on the rtcpeerconnection.
... syntax rtcpeerconnection.removestream(mediastream); parameters mediastream a mediastream specifying the stream to remove from the connection.
... example var pc, videostream; navigator.getusermedia({video: true}, function(stream) { pc = new rtcpeerconnection(); videostream = stream; pc.addstream(stream); } document.getelementbyid("closebutton").addeventlistener("click", function(event) { pc.removestream(videostream); pc.close(); }, false); ...
RTCPeerConnection: removestream event - Web APIs
the obsolete removestream event was sent to an rtcpeerconnection to inform it that a mediastream had been removed from the connection.
... you can use the rtcpeerconnection interface's onremovestream property to set a handler for this event.
... bubbles no cancelable no interface mediastreamevent event handler property rtcpeerconnection.onremovestream important: this event has been removed from the webrtc specification in favor of the existing removetrack event on the remote mediastream and the corresponding mediastream.onremovetrack event handler property of the remote mediastream.
... the rtcpeerconnection api is now track-based, so having zero tracks in the remote stream is equivalent to the remote stream being removed, which caused a removestream event.
RTCPeerConnection.sctp - Web APIs
the read-only sctp property on the rtcpeerconnection interface returns an rtcsctptransport describing the sctp transport over which sctp data is being sent and received.
... the sctp transport is used for transmitting and receiving data for any and all rtcdatachannels on the peer connection.
... syntax var sctp = rtcpeerconnection.sctp; value a rtcsctptransport object describing the sctp transport being used by the rtcpeerconnection for transmitting and receiving on its data channels, or null if sctp negotiation hasn't happened.
... example var pc = new rtcpeerconnection(); var channel = pc.createdatachannel("mydata"); channel.onopen = function(event) { channel.send('sending a message'); } channel.onmessage = function(event) { console.log(event.data); } // determine the largest message size that can be sent var sctp = pc.sctp; var maxmessagesize = sctp.maxmessagesize; specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.sctp' in that specification.
RTCPeerConnection: track event - Web APIs
the track event is sent to the ontrack event handler on rtcpeerconnections after a new track has been added to an rtcrtpreceiver which is part of the connection.
... bubbles yes cancelable no interface rtctrackevent event handler property ontrack by the time this event is delivered, the new track has been fully added to the peer connection.
... examples this example shows code that creates a new rtcpeerconnection, then adds a new track event handler.
... pc = new rtcpeerconnection({ iceservers: [ { urls: "turn:fake.turnserver.url", username: "someusername", credential: "somepassword" } ] }); pc.addeventlistener("track", e => { videoelement.srcobject = e.streams[0]; hangupbutton.disabled = false; }, false); the event handler assigns the new track's first stream to an existing <video> element, identified using the variable videoelement.
MIDIConnectionEvent - Web APIs
the midiconnectionevent interface of the web midi api is the event passed to the onstatechange event handler of the midiaccess interface and the onstatechange event of the midiports interface.
... constructor midiconnectionevent.midiconnectionevent creates a new midiconnectionevent object.
... properties midiconnectionevent.port returns a reference to a midiport instance for a port that has been connected or disconnected." examples specifications specification status comment web midi api working draft initial definition.
PerformanceTiming.secureConnectionStart - Web APIs
the legacy performancetiming.secureconnectionstart read-only property returns an unsigned long long representing the moment, in miliseconds since the unix epoch, where the secure connection handshake starts.
... if no such connection is requested, it returns 0.
... syntax time = performancetiming.secureconnectionstart; specifications specification status comment navigation timingthe definition of 'performancetiming.secureconnectionstart' in that specification.
RTCPeerConnection: datachannel event - Web APIs
a datachannel event is sent to an rtcpeerconnection instance when an rtcdatachannel has been added to the connection, as a result of the remote peer calling rtcpeerconnection.createdatachannel().
... note: this event is not dispatched when the local end of the connection creates the channel.
... this same code can also instead use the rtcpeerconnection interface's ondatachannel event handler property, like this: pc.ondatachannel = ev => { receivechannel = ev.channel; receivechannel.onmessage = myhandlemessage; receivechannel.onopen = myhandleopen; receivechannel.onclose = myhandleclose; } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'datacha...
RTCPeerConnection.getDefaultIceServers() - Web APIs
the getdefaulticeservers() method of the rtcpeerconnection interface returns an array of objects based on the rtciceserver dictionary, which indicates what, if any, ice servers the browser will use by default if none are provided to the rtcpeerconnection in its rtcconfiguration.
... syntax var defaulticeservers = rtcpeerconnection.getdefaulticeservers(); return value an array of ice servers, specified as objects based on rtciceserver, which the browser will use if none are specified in the configuration of the rtcpeerconnection.
... example var pc = new rtcpeerconnection(); var iceservers = pc.getdefaulticeservers(); if (iceservers.length === 0) { // deal with the lack of default ice servers, possibly by using our own defaults } specifications specification status comment webrtc extensions ...
RTCPeerConnection: idpassertionerror event - Web APIs
an idpassertionerror event informs the target, a rtcpeerconnection object, that the identity provider (idp) encountered an error when trying to generate an identity assertion.
... an event handler for this event can be added using the rtcpeerconnection.onidpassertionerror property.
... bubbles no cancelable no interface rtcidentityerrorevent event handler property onidpassertionerror warning: this event is no longer used; instead, you can detect an assertion error by detecting when the promise returned by rtcpeerconnection.peeridentity is rejected.
RTCPeerConnection: idpvalidationerror event - Web APIs
an idpvalidationerror event informs the target, a rtcpeerconnection object's onidpvalidationerror event handler, that the identity provider (idp) encountered an error while validating an identity assertion.
... an event handler for this event can be added using the rtcpeerconnection.onidpvalidationerror property.
... bubbles no cancelable no interface rtcidentityerrorevent event handler property onidpvalidationerror warning: this event is no longer used; instead, you can detect a validation error by detecting when the promise returned by rtcpeerconnection.peeridentity is rejected.
RTCPeerConnection.localDescription - Web APIs
the read-only property rtcpeerconnection.localdescription returns an rtcsessiondescription describing the session for the local end of the connection.
... syntax var sessiondescription = peerconnection.localdescription; on a more fundamental level, the returned value is the value of rtcpeerconnection.pendinglocaldescription if that property isn't null; otherwise, the value of rtcpeerconnection.currentlocaldescription is returned.
... var pc = new rtcpeerconnection(); … var sd = pc.localdescription; if (sd) { alert("local session: type='" + sd.type + "'; sdp description='" + sd.sdp + "'"); } else { alert("no local session yet."); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.localdescription' in that specification.
RTCPeerConnection.onicecandidateerror - Web APIs
the rtcpeerconnection.onicecandidateerror property is an eventhandler which specifies a function which is called to handle the icecandidateerror event when it occurs on an rtcpeerconnection instance.
... syntax rtcpeerconnection.onicecandidateerror = eventhandler; value this should be set to a function you provide which is passed a single parameter: an rtcpeerconnectioniceerrorevent object describing the icecandidateerror event.
... } } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.onicecandidateerror' in that specification.
RTCPeerConnection.onidpassertionerror - Web APIs
the rtcpeerconnection.onidpassertionerror event handler is a property containing the code to execute whent the idpassertionerror event, of type rtcidentityerrorevent, is received by this rtcpeerconnection.
...you should instead detect idp assertion errors by handling rejection of the promise returned by rtcpeerconnection.peeridentity.
... syntax peerconnection.onidpassertionerror = function; values function is the name of a user-defined function, without the () suffix or any parameters, or an anonymous function declaration, such as function(event) {...}.
RTCPeerConnection.onidpvalidationerror - Web APIs
the rtcpeerconnection.onidpvalidationerror event handler is a property containing the code to execute whent the idpvalidationerror event, of type rtcidentityerrorevent, is received by this rtcpeerconnection.
...you should instead detect idp validation errors by watching for the promise returned by rtcpeerconnection.peeridentity to be rejected.
... syntax peerconnection.onidpvalidationerror = function; values function is the name of a user-defined function, without the () suffix or any parameters, or an anonymous function declaration, such as function(event) {...}.
RTCPeerConnection: idpvalidationerror event - Web APIs
an idpvalidationerror event informs the target, a rtcpeerconnection object, that the identity provider (idp) encountered an error when trying to validate an identity assertion.
... an event handler for this event can be added via the rtcpeerconnection.onidpvalidationerror property.
...instead, you should watch for the rtcpeerconnection.peeridentity promise to be rejected with an operationerror.
WorkerNavigator.connection - Web APIs
the workernavigator.connection read-only property returns a networkinformation object containing information about the system's connection, such as the current bandwidth of the user's device or whether the connection is metered.
... this could be used to select high definition content or low definition content based on the user's connection.
... syntax connectioninfo = self.navigator.connection specifications specification status comment network information apithe definition of 'navigator.connection' in that specification.
RTCPeerConnection.onidentityresult - Web APIs
the rtcpeerconnection.onidentityresult event handler is a property containing the code to execute when the identityresult event, of type rtcidentityevent, is received by this rtcpeerconnection.
... syntax peerconnection.onidentityresult = function; values function is the name of a user-defined function, without the () suffix or any parameters, or an anonymous function declaration, such as function(event) {...}.
RTCPeerConnection.onpeeridentity - Web APIs
the rtcpeerconnection.onpeeridentity event handler is a property containing the code to execute whent the peeridentity event, of type event, is received by this rtcpeerconnection.
... syntax peerconnection.onpeeridentity = function; values function is the name of a user-defined function, without the () suffix or any parameters, or an anonymous function declaration, such as function(event) {...}.
RTCPeerConnection.setIdentityProvider() - Web APIs
the rtcpeerconnection.setidentityprovider() method sets the identity provider (idp) to the triplet given in parameter: its name, the protocol used to communicate with it (optional) and an optional username.
... example var pc = new peerconnection(); pc.setidentityassertion("developer.mozilla.org"); specifications specification status comment identity for webrtcthe definition of 'rtcpeerconnection.setidentityprovider()' in that specification.
RTCPeerConnection: signalingstatechange event - Web APIs
an signalingstatechange event is sent to an rtcpeerconnection to notify it that its signaling state, as indicated by the signalingstate property, has changed.
... bubbles no cancelable no interface event event handler property rtcpeerconnection.onsignalingstatechange examples given an rtcpeerconnection, pc, and an updatestatus() function that presents status information to the user, this code sets up an event handler to let the user know when the ice negotiation process finishes up.
RTCPeerConnectionIceEvent.candidate - Web APIs
the read-only candidate property of the rtcpeerconnectioniceevent interface returns the rtcicecandidate associated with the event.
... example pc.onicecandidate = function( ev ) { alert("the ice candidate (transport address: '" + ev.candidate.candidate + "') has been added to this connection."); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnectioniceevent.candidate' in that specification.
RTCPeerConnection: identityresult event - Web APIs
an identityresult event is sent to an rtcpeerconnection object's onidentityresult event handler to inform it that an assertion has been generated by an associated identity provider (idp) during the process of creating an sdp offer or answer.
Index - Web APIs
WebAPIIndex
this stream can be used in a similar way as a mediastream obtained via navigator.getusermedia — it can, for example, be sent to a remote peer using the rtcpeerconnection addstream() method.
...examples include: 157 audionode.channelcount api, audionode, property, reference, web audio api, channelcount the channelcount property of the audionode interface represents an integer used to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node.
... 365 broadcastchannel.close() api, broadcast channel api, broadcastchannel, experimental, html api, method, reference the broadcastchannel.close() terminates the connection to the underlying channel, allowing the object to be garbage collected.
...And 242 more matches
Signaling and video calling - Web APIs
a connection is established through a discovery and negotiation process called signaling.
... note: if you try out the example on glitch, please note that any changes made to the code will immediately reset any connections.
... the signaling server establishing a webrtc connection between two devices requires the use of a signaling server to resolve how to connect them over the internet.
...And 73 more matches
Using DTMF with WebRTC - Web APIs
in order to more fully support audio/video conferencing, webrtc supports sending dtmf to the remote peer on an rtcpeerconnection.
... this article offers a brief high-level overview of how dtmf works over webrtc, then provides a guide for everyday developers about how to send dtmf over an rtcpeerconnection.
... sending dtmf on an rtcpeerconnection a given rtcpeerconnection can have multiple media tracks sent or received on it.
...And 48 more matches
sslfnc.html
if client does not provide certificate, the connection terminates.
...if it is on, then pr_accept configures the ssl socket to handshake as a client, even though it accepted the connection as a tcp server.
...if you turn this option off, an attempt to establish a connection with a peer that only understands ssl v3 will fail.
...And 39 more matches
WebRTC API - Web APIs
the documentation you'll find here will help you understand the fundamentals of webrtc, how to set up and use both data and media connections, and more.
...connections between peers can be made without requiring any special drivers or plug-ins, and can often be made without any intermediary servers.
... connections between two peers are represented by the rtcpeerconnection interface.
...And 36 more matches
A simple RTCDataChannel sample - Web APIs
in this example, we will open an rtcdatachannel connection linking two elements on the same page.
...we'll cover the mechanics of accomplishing the connection and transmitting and receiving data, but we will save the bits about locating and linking to a remote computer for another example.
...first, we have a couple of buttons for establishing and closing the connection: <button id="connectbutton" name="connectbutton" class="buttonleft"> connect </button> <button id="disconnectbutton" name="disconnectbutton" class="buttonright" disabled> disconnect </button> then there's a box which contains the text input box into which the user can type a message to transmit, with a button to send the entered text.
...And 31 more matches
Sqlite.jsm
there is even a shrinkmemory api that will minimize memory usage of the connection automatically.
... before you can use this module, you need to import it into your scope: components.utils.import("resource://gre/modules/sqlite.jsm") obtaining a connection sqlite.jsm exports the sqlite symbol.
... this symbol is an object with a single function: openconnection.
...And 30 more matches
WebRTC connectivity - Web APIs
this article describes how the various webrtc-related protocols interact with one another in order to create a connection and transfer data and/or media among peers.
... signaling unfortunately, webrtc can’t create connections without some sort of server in the middle.
...it’s any sort of channel of communication to exchange information before setting up a connection, whether by email, post card or a carrier pigeon...
...And 29 more matches
mozIStorageService
storage/public/mozistorageservice.idlscriptable this interface lets you open a mozistorageconnection to a database file, as well as create backups of an unopened database file.
... this is the only way to open a database connection.
... 1.0 66 introduced gecko 1.8 inherits from: nsisupports last changed in gecko 1.9 (firefox 3) see mozistorageconnection method overview nsifile backupdatabasefile(in nsifile adbfile, in astring abackupfilename, [optional] in nsifile abackupparentdirectory); mozistorageconnection opendatabase(in nsifile adatabasefile); mozistorageconnection openspecialdatabase(in string astoragekey); mozistorageconnection openunshareddatabase(in nsifile adatabasefile); methods backupdatabasefile() this method makes a backup of the specified file.
...And 22 more matches
Index - MDN Web Docs Glossary: Definitions of Web-related terms
4 alpn alpn, draft, glossary, needscontent, tls application-layer protocol negotiation (alpn) is a tls extension which indicates what application layer protocol is negotiating the encryped connection without requiring additional round trips.
... 26 bandwidth glossary, infrastructure bandwidth is the measure of how much information can pass through a data connection in a given amount of time.
... 120 domain sharding dns, glossary, http, reference, web performance, latency browsers limit the number of active connections for each domain.
...And 20 more matches
Index
that's a good opportunity to talk about ssl/tls connections to servers in general (not just ev, not just websites).
...(tls is a newer version of ssl with enhanced features.) when establishing an ssl connection to a server, (at least) a server certificate (and its trust chain) is exchanged from the server to the client (e.g., the browser), and the client verifies that the certificate can be verified (including matching the name of the expected destination server).
...the combination of the hash and encryption algorithms used for a ssl connection is called a cipher suite.
...And 18 more matches
Storage
open a connection to the database of your choice - mozistorageconnection.
... create statements to execute on the connection - mozistoragestatement.
... opening a connection javascript example of opening my_db_file_name.sqlite in the profile directory: components.utils.import("resource://gre/modules/services.jsm"); components.utils.import("resource://gre/modules/fileutils.jsm"); let file = fileutils.getfile("profd", ["my_db_file_name.sqlite"]); let dbconn = services.storage.opendatabase(file); // will also create the file if it does not exist likewise, the c++ would look like this: nscomptr<nsifile> dbfile; rv = ns_getspecialdirectory(ns_app_user_profile_50_dir, getter_addrefs(dbfile)); ns_ensure_success(rv, rv); rv = dbfile->append(ns_literal_string("my_db_file_name.sqlite")); ns_ensure_success(rv, rv);...
...And 17 more matches
Navigation and resource timings - Web Performance
as displayed in the image below, the navigation process goes from navigationstart, unloadeventstart, unloadeventend, redirectstart, redirectend, fetchstart, domainlookupstart, domainlookupend, connectstart , connectend, secureconnectionstart, requeststart, responsestart, responseend, domloading, dominteractive, domcontentloadedeventstart, domcontentloadedeventend, domcomplete, loadeventstart, and loadeventend.
... with the metrics above, and a little bit of math, we can calculate many important metrics like time to first byte, page load time, dns lookup, and whether the connection is secure.
... secureconnectionstart when the secure connection handshake starts.
...And 17 more matches
Autodial for Windows NT - Archive of obsolete content
a brief history of autodial on windows in the consumer versions of windows, (windows 95, windows 98, windows me) autodial for all applications is controlled from the control panel, in the internet options applet, under the tab connections.
...microsoft added a windows system service (remote access auto connection) to handle autodial for all other applications.
... if this service is running, and control panel | network connections | advanced | dialup preferences is set to enable autodial, then any application trying to access the internet will trigger the autodial feature whenever an internet address cannot be reached.
...And 16 more matches
Protocol upgrade mechanism - HTTP
the http/1.1 protocol provides a special mechanism that can be used to upgrade an already established connection to a different protocol, using the upgrade header field.
...implementations can choose not to take advantage of an upgrade even if they support the new protocol, and in practice, this mechanism is used mostly to bootstrap a websockets connection.
... upgrading http/1.1 connections the upgrade header field is used by clients to invite the server to switch to one of the listed protocols, in descending preference order.
...And 16 more matches
Lifetime of a WebRTC session - Web APIs
in this article, we'll look at the lifetime of a webrtc session, from establishing the connection all the way through closing the connection when it's no longer needed.
... this article doesn't get into details of the actual apis involved in establishing and handling a webrtc connection; it simply reviews the process in general with some information about why each step is required.
... establishing the connection the internet is big.
...And 14 more matches
Necko walkthrough
then in necko http code (still on the main thread for now): nshttpchannel::asyncopen nshttpchannel::beginconnect() creates nshttpconnectioninfo object for the channel checks if we're proxying or not fires off the dns prefetch request (dispatched to dns thread pool) some other things nshttpchannel::connect might to a speculativeconnect (pre open tcp socket) nshttpchannel::continueconnect some cache stuff nshttpchannel::setuptransaction creates new nshttptransaction, and inits it with mreque...
...eaders) and muploadstream (which was created from the request data in channel setup) gets an nsiasyncinputstream (for the response; corresponds to the nspipeinputstream for the response stream pipe) passes it to nsinputstreampump nshttpchannel::ghttphandler->initiatetransaction (called from connect) this is the global nshttphandler object, which adds the transaction to the nshttpconnectionmgr (one of these per nshttphandler).
... nshttpconnectionmgr::postevent creates an nsconnevent with params including the handler function, nshttpconnectionmgr::onmsgnewtransaction, and the recently created nshttptransaction.
...And 13 more matches
CloseEvent - Web APIs
a closeevent is sent to clients using websockets when the connection is closed.
... 1000 normal closure normal closure; the connection successfully completed whatever purpose for which it was created.
... 1001 going away the endpoint is going away, either because of a server failure or because the browser is navigating away from the page that opened the connection.
...And 13 more matches
RTCConfiguration - Web APIs
the rtcconfiguration dictionary is used to provide configuration options for an rtcpeerconnection.
... it may be passed into the constructor when instantiating a connection, or used with the rtcpeerconnection.getconfiguration() and rtcpeerconnection.setconfiguration() methods, which allow inspecting and changing the configuration while a connection is established.
... certificates optional an array of objects of type rtccertificate which are used by the connection for authentication.
...And 13 more matches
WebRTC Statistics API - Web APIs
most broadly, you can call getstats() on an rtcpeerconnection to get statistics for the connection overall.
... in this example, a new rtcpeerconnection is created, and then setinterval() is used to set the function getconnectionstats() to be called every second.
... that function, in turn, uses getstats() to obtain statistics for the connection and to make use of that data.
...And 13 more matches
NSS Tools ssltap
it watches tcp connections and displays the data going by.
... if a connection is ssl, the data display includes interpreted ssl records and handshaking.
... description the ssltap command opens a socket on a rendezvous port and waits for an incoming connection from the client side.
...And 12 more matches
RTCDtlsTransport - Web APIs
the rtcdtlstransport interface provides access to information about the datagram transport layer security (dtls) transport over which a rtcpeerconnection's rtp and rtcp packets are sent and received by its rtcrtpsender and rtcrtpreceiver objects.
... a dtls transport is also used to provide information about sctp packets transmitted and received by an connection's data channels.
...the number of dtls transports created and how they're used depends on the bundling mode used when creating the rtcpeerconnection.
...And 11 more matches
Writing WebSocket client applications - Web APIs
creating a websocket object in order to communicate using the websocket protocol, you need to create a websocket object; this will automatically attempt to open the connection to the server.
...this should use the url scheme wss://, although some software may allow you to use the insecure ws:// for local connections.
...this may happen if you attempt to use an insecure connection (most user agents now require a secure link for all websocket connections unless they're on the same device or possibly on the same network).
...And 11 more matches
An overview of HTTP - HTTP
WebHTTPOverview
it is an application layer protocol that is sent over tcp, or over a tls-encrypted tcp connection, though any reliable transport protocol could theoretically be used.
... http is stateless, but not sessionless http is stateless: there is no link between two requests being successively carried out on the same connection.
... http and connections a connection is controlled at the transport layer, and therefore fundamentally out of scope for http.
...And 11 more matches
Codecs used by WebRTC - Web media technologies
this is due to a change in google play store requirements that prevent firefox from downloading and installing the openh264 codec needed to handle h.264 in webrtc connections.
... other video codecs codec name profile(s) browser compatibility vp9 — chrome (48+), firefox vp8 vp8, which we describe in general in the main guide to video codecs used on the web, has some specific requirements that must be followed when using it to encode or decode a video track on a webrtc connection.
... parameters which must be specified these parameters must be specified whenever using avc in a webrtc connection.
...And 11 more matches
Introduction to SSL - Archive of obsolete content
it uses tcp/ip on behalf of the higher-level protocols, and in the process allows an ssl-enabled server to authenticate itself to an ssl-enabled client, allows the client to authenticate itself to the server, and allows both machines to establish an encrypted connection.
... an encrypted ssl connection requires all information sent between a client and a server to be encrypted by the sending software and decrypted by the receiving software, thus providing a high degree of confidentiality.
...in addition, all data sent over an encrypted ssl connection is protected with a mechanism for detecting tampering-that is, for automatically determining whether the data has been altered in transit.
...And 10 more matches
nsISocketTransport
inherits from: nsitransport last changed in gecko 6.0 (firefox 6.0 / thunderbird 6.0 / seamonkey 2.3) note: connection setup is triggered by opening an input or output stream, it does not start on its own.
... completion of the connection setup is indicated by a status_connected_to notification to the event sink (if set).
... unsigned long gettimeout(in unsigned long atype); boolean isalive(); void settimeout(in unsigned long atype, in unsigned long avalue); attributes attribute type description connectionflags unsigned long a bitmask that can be used to modify underlying behavior of the socket connection.
...And 10 more matches
PerformanceTiming - Web APIs
if a persistent connection is used, or the information is stored in a cache or a local resource, the value will be the same as performancetiming.fetchstart.
...if a persistent connection is used, or the information is stored in a cache or a local resource, the value will be the same as performancetiming.fetchstart.
... performancetiming.connectstart read only when the request to open a connection is sent to the network.
...And 10 more matches
Writing WebSocket servers - Web APIs
the websocket handshake first, the server must listen for incoming socket connections using a standard tcp socket.
...browsers generally require a secure connection for websockets, although they may offer an exception for local devices.
...in the handshake, details of the connection are negotiated, and either party can back out before completion if the terms are unfavorable.
...And 10 more matches
HTTP Index - HTTP
WebHTTPIndex
http follows a classical client-server model, with a client opening a connection to make a request, then waiting until it receives a response.
... 15 connection management in http/1.x connection management, guide, http, networking, performance, webmechanics connection management is a key topic in http: opening and maintaining connections largely impacts the performance of web sites and web applications.
... in http/1.x, there are several models: short-lived connections, persistent connections, and http pipelining.
...And 10 more matches
nsISocketProvider
anonymous_connect 1 << 1 when setting this flag, the socket will not apply any credentials when establishing a connection.
... for example, an ssl connection would not send any client-certificates if this flag is set.
... ahost the hostname for this connection.
...And 9 more matches
nsIWebSocketChannel
originaluri nsiuri the original uri used to construct the protocol connection.
... uri nsiuri the uri corresponding to the protocol connection after any redirections are completed.
... 1000 close_normal normal closure; the connection successfully completed whatever purpose for which it was created.
...And 9 more matches
Understanding latency - Web Performance
in terms of performance optimization, it's important to optimize to reduce causes of lacency and to test site performance emulating high latency to optimizer for users with lousy connections.
...subsequent requests will have less latency because the connection to the server is already set.
... latency describes the amount of delay on a network or internet connection.
...And 9 more matches
NSS tools : ssltab
name ssltap — tap into ssl connections and display the data going by synopsis libssltap [-vhfsxl] [-p port] [hostname:port] description the ssl debugging tool ssltap is an ssl-aware command-line proxy.
... it watches tcp connections and displays the data going by.
... if a connection is ssl, the data display includes interpreted ssl records and handshaking options -v print a version string for the tool.
...And 8 more matches
NSS tools : ssltap
name ssltap — tap into ssl connections and display the data going by synopsis libssltap [-vhfsxl] [-p port] [hostname:port] description the ssl debugging tool ssltap is an ssl-aware command-line proxy.
... it watches tcp connections and displays the data going by.
... if a connection is ssl, the data display includes interpreted ssl records and handshaking options -v print a version string for the tool.
...And 8 more matches
NSS tools : ssltap
MozillaProjectsNSStoolsssltap
name ssltap — tap into ssl connections and display the data going by synopsis libssltap [-vhfsxl] [-p port] [hostname:port] description the ssl debugging tool ssltap is an ssl-aware command-line proxy.
... it watches tcp connections and displays the data going by.
... if a connection is ssl, the data display includes interpreted ssl records and handshaking options -v print a version string for the tool.
...And 8 more matches
nsINetworkLinkService
by: @mozilla.org/network/network-link-service;1 as a service: var networklinkservice = components.classes["@mozilla.org/network/network-link-service;1"] .getservice(components.interfaces.nsinetworklinkservice); attributes attribute type description islinkup boolean this is set to true when the system is believed to have a usable network connection.
... the link is only up when network connections can be established.
... linktype unsigned long the type of network connection, one of the link_type_* constants.
...And 8 more matches
EventSource - Web APIs
an eventsource instance opens a persistent connection to an http server, which sends events in text/event-stream format.
... the connection remains open until closed by calling eventsource.close().
... once the connection is opened, incoming messages from the server are delivered to your code in the form of events.
...And 8 more matches
RTCIceCandidateStats.networkType - Web APIs
the string indicates the type of network connection that the described candidate would use to communicate with the other peer.
... the permitted values are: bluetooth a bluetooth connection is used by the described connection.
... cellular the connection uses a cellular data service to connect.
...And 8 more matches
Using WebRTC data channels - Web APIs
once you've established a webrtc peer connection using the rtcpeerconnection interface, you're able to send and receive media data between the two peers on the connection.
...in this guide, we'll examine how to add a data channel to a peer connection, which can then be used to securely exchange arbitrary data; that is, any kind of data we wish, in any format we choose.
... automatic negotiation often, you can allow the peer connection to handle negotiating the rtcdatachannel connection for you.
...And 8 more matches
WebSocket - Web APIs
WebAPIWebSocket
the websocket object provides the api for creating and managing a websocket connection to a server, as well as for sending and receiving data on the connection.
... constants constant value websocket.connecting 0 websocket.open 1 websocket.closing 2 websocket.closed 3 properties websocket.binarytype the binary data type used by the connection.
... websocket.onclose an event listener to be called when the connection is closed.
...And 8 more matches
Index
MozillaTechXPCOMIndex
objects implementing this interface can be registered with mozistorageconnection.createaggregatefunction().
... 208 mozistorageconnection interfaces, storage, xpcom, xpcom api reference for a general overview on how to use this interface, see storage.
... objects implementing this interface can be registered with mozistorageconnection.createfunction().
...And 7 more matches
Network Information API - Web APIs
the network information api provides information about the system's connection in terms of general connection type (e.g., 'wifi', 'cellular', etc.).
... this can be used to select high definition content or low definition content based on the user's connection.
... the entire api consists of the addition of the networkinformation interface and a single property to the navigator interface: navigator.connection.
...And 7 more matches
RTCNetworkType - Web APIs
the webrtc rtcnetworktype enumerated type defines a set of strings used to identify the type of network used by a connection between two peers.
... this type is used as the value or the following properties: rtcicecandidate's networktype rtcstunserverconnectionstats's networktype values bluetooth a bluetooth connection is used by the described connection.
... cellular the connection uses a cellular data service to connect.
...And 7 more matches
RTCTrackEvent - Web APIs
the webrtc api interface rtctrackevent represents the track event, which is sent when a new mediastreamtrack is added to an rtcrtpreceiver which is part of the rtcpeerconnection.
... the target is the rtcpeerconnection object to which the track is being added.
...you will probably not need to create new track events yourself, since they're typically created by the webrtc infrastructure and sent to the connection's ontrack event handler.
...And 7 more matches
Introduction to the Real-time Transport Protocol (RTP) - Web APIs
those conditions may be affected by everything from the underlying layers of the network stack to the physical network connection, the intervening networks, the performance of the remote endpoint, noise levels, traffic levels, and so forth.
... rtcpeerconnection and rtp each rtcpeerconnection has methods which provide access to the list of rtp transports that service the peer connection.
... these correspond to the following three types of transport supported by rtcpeerconnection: rtcrtpsender rtcrtpsenders handle the encoding and transmission of mediastreamtrack data to a remote peer.
...And 7 more matches
Populating the page: how browsers work - Web Performance
tcp handshake once the ip address is known, the browser sets up a connection to the server via a tcp three-way handshake.
... this mechanism is designed so that two entities attempting to communicate—in this case the browser and web server—can negotiate the parameters of the network tcp socket connection before transmitting data, often over https.
... tls negotiation for secure connections established over https, another "handshake" is required.
...And 7 more matches
RTCConfiguration.bundlePolicy - Web APIs
the rtcconfiguration dictionary's bundlepolicy property is a string value indicating which sdp bundling policy, if any, to use for the underlying rtp streams used by an rtcpeerconnection.
... the configuration object is used as an input to the rtcpeerconnection() constructor.
... syntax let rtcconfiguration = { bundlepolicy: policy }; rtcconfiguration.bundlepolicy = policy; value a domstring identifying the sdp bundling policy to use for the rtp streams used by the rtcpeerconnection.
...And 6 more matches
RTCIceCandidatePairStats - Web APIs
the webrtc rtcicecandidatepairstats dictionary reports statistics which provide insight into the quality and performance of an rtcpeerconnection while connected and configured as described by the specified pair of ice candidates.
... circuitbreakertriggercount optional an integer value indicating the number of times the circuit-breaker has been triggered for this particular 5-tuple (the set of five values comprising a tcp connection: source ip address, source port number, destination ip address, destination port number, and protocol).
... the circuit breaker is triggered whenever a connection times out or otherwise needs to be automatically aborted.
...And 6 more matches
RTCOfferOptions.iceRestart - Web APIs
the icerestart property of the rtcofferoptions dictionary is a boolean value which, when true, tells the rtcpeerconnection to start ice renegotiation.
... note: rather than manually creating and submitting an offer with icerestart set to true, you should consider calling the rtcpeerconnection method restartice() instead.
... syntax var options = { icerestart: trueorfalse }; value a boolean value indicating whether or not the rtcpeerconnection should generate new values for the connection's ice-ufrag and ice-pwd values, which will trigger ice renegotiation on the next message sent to the remote peer.
...And 6 more matches
RTCStatsReport - Web APIs
the rtcstatsreport interface provides a statistics report obtained by calling one of the rtcpeerconnection.getstats(), rtcrtpreceiver.getstats(), and rtcrtpsender.getstats() methods.
... calling getstats() on an rtcpeerconnection lets you specify whether you wish to obtain statistics for outbound, inbound, or all streams on the connection.
...candidate pairs other than the currently active pair for the transport are deleted when the rtcpeerconnection changes its rtcpeerconnection.icegatheringstate to new during an ice restart.
...And 6 more matches
Using server-sent events - Web APIs
this is one-way connection, so you can't send events from a client to a server.
... receiving events from the server the server-sent event api is contained in the eventsource interface; to open a connection to the server to begin receiving events from it, create a new eventsource object with the url of a script that generates the events.
... when not used over http/2, sse suffers from a limitation to the maximum number of open connections, which can be especially painful when opening multiple tabs, as the limit is per browser and is set to a very low number (6).
...And 6 more matches
Introduction to WebRTC protocols - Web APIs
there are many reasons why a straight up connection from peer a to peer b simply won’t work.
... it needs to bypass firewalls that would prevent opening connections, give you a unique address if like most situations your device doesn’t have a public ip address, and relay data through a server if your router doesn’t allow you to directly connect with peers.
... stun session traversal utilities for nat (stun) (acronym within an acronym) is a protocol to discover your public address and determine any restrictions in your router that would prevent a direct connection with a peer.
...And 6 more matches
Event reference
websocket events event name fired when open a websocket connection has been established.
... error a websocket connection has been closed with prejudice (some data couldn't be sent for example).
... close a websocket connection has been closed.
...And 6 more matches
PR_Accept
accepts a connection on a specified socket.
... syntax #include <prio.h> prfiledesc* pr_accept( prfiledesc *fd, prnetaddr *addr, printervaltime timeout); parameters the function has the following parameters: fd a pointer to a prfiledesc object representing the rendezvous socket on which the caller is willing to accept new connections.
... returns the function returns one of the following values: upon successful acceptance of a connection, a pointer to a new prfiledesc structure representing the newly accepted connection.
...And 5 more matches
mozIStorageFunction
objects implementing this interface can be registered with mozistorageconnection.createfunction().
...this is called by the database engine when the function registered with mozistorageconnection.createfunction() is used in an executing sql statement or trigger.
...if you use mozistorageconnection.executeasync() or, mozistoragestatement.executeasync() this callback will run on a different thread from the rest of your code.
...And 5 more matches
nsIServerSocket
the nsiserversocket interface implements a server socket that can accept incoming connections.
...it will asynchronously listen for and accept client connections.
... the listener will be notified once for each client connection that is accepted.
...And 5 more matches
Network request details - Firefox Developer Tools
"name": "x-varnish", "value": "766019457, 417549316" } ] }, "request headers (665 b)": { "headers": [ { "name": "accept", "value": "*/*" }, { "name": "accept-encoding", "value": "gzip, deflate, br" }, { "name": "accept-language", "value": "en-us,en;q=0.5" }, { "name": "connection", "value": "keep-alive" }, { "name": "cookie", "value": "wmf-last-access=11-jun-2019; wmf-last-access-global=11-jun-2019; mwphp7seed=5c9; geoip=us:ny:port_jervis:41.38:-74.67:v4" }, { "name": "dnt", "value": "1" }, { "name": "host", "value": "en.wikipedia.org" }, { "name": "referer", ...
...if the response is html, js, or css, it will be shown as text: if the response is json, it will be shown as an inspectable object: if the response is an image, the tab displays a preview: for network responses that are initiated by a websocket connection, the details pane shows any associated messages.
... request timing the request timing section breaks a network request down into the following subset of the stages defined in the http archive specification: name description blocked time spent in a queue waiting for a network connection.
...And 5 more matches
RTCConfiguration.certificates - Web APIs
the rtcconfiguration dictionary's optional certificates property is an array of rtccertificate objects providing the security certificates available for use when authenticating duing the connection process.
...if this property isn't specified, the browser will automatically generate and use a certificate to secure the connection.
... description if this property isn't included in the configuration, a set of certificates is automatically generated for each instance of rtcpeerconnection.
...And 5 more matches
Writing a WebSocket server in C# - Web APIs
this server conforms to rfc 6455 so it will only handle connections from chrome version 16, firefox 11, ie 10 and over.
... first steps websockets communicate over a tcp (transmission control protocol) connection.
... methods: start() system.net.sockets.tcpclient accepttcpclient() waits for a tcp connection, accepts it and returns it as a tcpclient object.
...And 5 more matches
Evolution of HTTP - HTTP
at this point, a typical request and response looked like this: get /mypage.html http/1.0 user-agent: ncsa_mosaic/2.0 (windows 3.1) 200 ok date: tue, 15 nov 1994 08:12:31 gmt server: cern/3.0 libwww/2.17 content-type: text/html <html> a page with an image <img src="/myimage.gif"> </html> followed by a second connection and request to fetch the image (followed by a response to that request): get /myimage.gif http/1.0 user-agent: ncsa_mosaic/2.0 (windows 3.1) 200 ok date: tue, 15 nov 1994 08:12:32 gmt server: cern/3.0 libwww/2.17 content-type: text/gif (image content) these novelties have not been introduced as concerted effort, but as a try-and-see approach over the 1991-1995 period: a server and a browser a...
... http/1.1 clarified ambiguities and introduced numerous improvements: a connection can be reused, saving the time to reopen it numerous times to display the resources embedded into the single original document retrieved.
... a typical flow of requests, all through one single connection is now looking like this: get /docs/glossary/simple_header http/1.1 host: developer.mozilla.org user-agent: mozilla/5.0 (macintosh; intel mac os x 10.9; rv:50.0) gecko/20100101 firefox/50.0 accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 accept-language: en-us,en;q=0.5 accept-encoding: gzip, deflate, br referer: https://developer.mozilla.org/docs/glossary/simple_header 2...
...And 5 more matches
Cross-Origin Resource Sharing (CORS) - HTTP
WebHTTPCORS
a “simple request” is one that meets all the following conditions: one of the allowed methods: get head post apart from the headers automatically set by the user agent (for example, connection, user-agent, or the other headers defined in the fetch spec as a “forbidden header name”), the only headers which are allowed to be manually set are those which the fetch spec defines as a “cors-safelisted request-header”, which are: accept accept-language content-language content-type (but note the additional requirements below) dpr downlink save-data viewport-width w...
...eges: let's look at what the browser will send to the server in this case, and let's see how the server responds: get /resources/public-data/ http/1.1 host: bar.other user-agent: mozilla/5.0 (macintosh; intel mac os x 10.14; rv:71.0) gecko/20100101 firefox/71.0 accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 accept-language: en-us,en;q=0.5 accept-encoding: gzip,deflate connection: keep-alive origin: https://foo.example the request header of note is origin, which shows that the invocation is coming from https://foo.example.
... http/1.1 200 ok date: mon, 01 dec 2008 00:23:53 gmt server: apache/2 access-control-allow-origin: * keep-alive: timeout=2, max=100 connection: keep-alive transfer-encoding: chunked content-type: application/xml […xml data…] in response, the server sends back an access-control-allow-origin header.
...And 5 more matches
Upgrade - HTTP
WebHTTPHeadersUpgrade
the http 1.1 (only) upgrade header can be used to upgrade an already established client/server connection to a different protocol (over the same transport protocol).
... for example, it can be used by a client to upgrade a connection from http 1.1 to http 2.0, or an http or https connection into a websocket.
... for example, the client might send a get request as shown, listing the preferred protocols to switch to (in this case "example/1" and "foo/2"): get /index.html http/1.1 host: www.example.com connection: upgrade upgrade: example/1, foo/2 note: connection: upgrade must be set whenever upgrade is sent.
...And 5 more matches
HTTP headers - HTTP
WebHTTPHeaders
headers can also be grouped according to how proxies handle them: connection keep-alive proxy-authenticate proxy-authorization te trailer transfer-encoding upgrade (see also protocol upgrade mechanism).
... hop-by-hop headers these headers are meaningful only for a single transport-level connection, and must not be retransmitted by proxies or cached.
... note that only hop-by-hop headers may be set using the connection general header.
...And 5 more matches
A typical HTTP session - HTTP
WebHTTPSession
in client-server protocols, like http, sessions consist of three phases: the client establishes a tcp connection (or the appropriate connection if the transport layer is not tcp).
... as of http/1.1, the connection is no longer closed after completing the third phase, and the client is now granted a further request: this means the second and third phases can now be performed any number of times.
... establishing a connection in client-server protocols, it is the client which establishes the connection.
...And 5 more matches
PR_NewTCPSocket
description tcp (transmission control protocol) is a connection-oriented, reliable byte-stream protocol of the tcp/ip protocol suite.
...a tcp connection is established by a passive socket (the server) accepting a connection setup request from an active socket (the client).
... typically, the server binds its socket to a well-known port with pr_bind, calls pr_listen to start listening for connection setup requests, and calls pr_accept to accept a connection.
...And 4 more matches
PR_OpenTCPSocket
description tcp (transmission control protocol) is a connection-oriented, reliable byte-stream protocol of the tcp/ip protocol suite.
...a tcp connection is established by a passive socket (the server) accepting a connection setup request from an active socket (the client).
... typically, the server binds its socket to a well-known port with pr_bind, calls pr_listen to start listening for connection setup requests, and calls pr_accept to accept a connection.
...And 4 more matches
NetworkInformation - Web APIs
the networkinformation interface provides information about the connection a device is using to communicate with the network and provides a means for scripts to be notified if the connection type changes.
...it is instead accessed through the connection property of the navigator interface.
... networkinformation.downlinkmax read only returns the maximum downlink speed, in megabits per second (mbps), for the underlying connection technology.
...And 4 more matches
RTCDataChannel.readyState - Web APIs
the read-only rtcdatachannel property readystate returns an enum of type rtcdatachannelstate which indicates the state of the data channel's underlying data connection.
... rtcdatachannelstate enum the rtcdatachannelstate enum defines string constants which reflect the current status of the rtcdatachannel's underlying data connection.
... constant description "connecting" the user agent (browser) is in the process of creating the underlying data transport; that is, whatever network level connection is used to link the two peers together is in the process of being set up.
...And 4 more matches
RTCDataChannel - Web APIs
every data channel is associated with an rtcpeerconnection, and each peer connection can have up to a theoretical maximum of 65,534 data channels (the actual limit may vary from browser to browser).
... to create a data channel and ask a remote peer to join you, call the rtcpeerconnection's createdatachannel() method.
... the peer being invited to exchange data receives a datachannel event (which has type rtcdatachannelevent) to let it know the data channel has been added to the connection.
...And 4 more matches
RTCDtlsTransport.state - Web APIs
connecting dtls is in the process of negotiating a secure connection and verifying the remote fingerprint.
... connected dtls has completed negotiation of a secure connection and verified the remote fingerprint.
... closed the transport has been closed intentionally as the result of receipt of a close_notify alert, or calling rtcpeerconnection.close().
...And 4 more matches
RTCIceServers.urls - Web APIs
WebAPIRTCIceServerurls
a single ice server this example creates a new rtcpeerconnection which will use a stun server at stunserver.example.org to negotiate connections.
... mypeerconnection = new rtcpeerconnection({ iceservers: [ { urls: "stun:stunserver.example.org" } ] }); notice that only the urls property is provided; the stun server doesn't require authentication, so this is all that's needed.
... a single ice server with authentication the second example creates a new rtcpeerconnection which will use a turn server at turnserver.example.org to negotiate connections.
...And 4 more matches
RTCIceTransport - Web APIs
this is particularly useful if you need to access state information about the connection.
...the value of state can be used to determine whether the ice agent has made an initial connection using a viable candidate pair ("connected"), made its final selection of candidate pairs ("completed"), or in an error state ("failed"), among other states.
...these are the same candidates which have already been sent to the remote peer by sending an icecandidate event to the rtcpeerconnection for transmission.
...And 4 more matches
RTCStatsType - Web APIs
candidate pairs other than the currently active pair for the transport are deleted when the rtcpeerconnection changes its rtcpeerconnection.icegatheringstate to new during an ice restart.
... codec an rtccodecstats object containing statistics about a codec currently being used by rtp streams to send or receive data for the rtcpeerconnection.
... data-channel an rtcdatachannelstats object which contains statistics about each rtcdatachannel on the connection.
...And 4 more matches
How to check the security state of an XMLHTTPRequest over SSL - Web APIs
the function is passed the channel property of an xmlhttprequest to extract the following information: was the connection secure?
... the channel property becomes available only after the request is sent and the connection was established, that is, on readystate loaded, interactive or completed.
...urityunsupportedtlsversionerror'; break; case 12: // ssl_error_bad_cert_domain, ssl(12) errname = 'securitycertificatedomainmismatcherror'; break; default: errname = 'securityerror'; break; } } } else { errtype = 'network'; switch (status) { // connect to host:port failed case 0x804b000c: // ns_error_connection_refused, network(13) errname = 'connectionrefusederror'; break; // network timeout error case 0x804b000e: // ns_error_net_timeout, network(14) errname = 'networktimeouterror'; break; // hostname lookup failed case 0x804b001e: // ns_error_unknown_host, network(30) errname = 'domainnotfounderror'; break; case 0x804b0047: ...
...And 4 more matches
Keep-Alive - HTTP
the keep-alive general header allows the sender to hint about how the connection may be used to set a timeout and a maximum amount of requests.
... the connection header needs to be set to "keep-alive" for this header to have any meaning.
... connection-specific header fields such as connection and keep-alive are prohibited in http/2.
...And 4 more matches
TCP handshake - MDN Web Docs Glossary: Definitions of Web-related terms
tcp (transmission control protocol) is a transport layer host-to-host protocol for connection-oriented communication between two computers on an ip network.
... tcp uses a three-way handshake (aka tcp-handshake, three message handshake, and/or syn-syn-ack) to set up a tcp/ip connection over an ip based network.
...the three message mechanism is designed so that two computers that want to pass information back and forth to each other can negotiate the parameters of the connection before transmitting data such as http browser requests.
...And 3 more matches
PR_Connect
initiates a connection on a specified socket.
... returns the function returns one of the following values: upon successful completion of connection setup, pr_success.
... if the socket is a tcp socket, pr_connect establishes a tcp connection to the peer.
...And 3 more matches
PR_Listen
listens for connections on a specified socket.
... syntax #include <prio.h> prstatus pr_listen( prfiledesc *fd, printn backlog); parameters the function has the following parameters: fd a pointer to a prfiledesc object representing a socket that will be used to listen for new connections.
... backlog the maximum length of the queue of pending connections.
...And 3 more matches
An overview of NSS Internals
that's a good opportunity to talk about ssl/tls connections to servers in general (not just ev, not just websites).
...(tls is a newer version of ssl with enhanced features.) when establishing an ssl connection to a server, (at least) a server certificate (and its trust chain) is exchanged from the server to the client (e.g., the browser), and the client verifies that the certificate can be verified (including matching the name of the expected destination server).
...the combination of the hash and encryption algorithms used for a ssl connection is called a cipher suite.
...And 3 more matches
sslintro.html
to avoid the overhead of repeating the full ssl handshake in situations like this, the ssl protocol supports the use of a session cache, which retains information about each connection, such as the master secret generated during the ssl handshake, for a predetermined length of time.
... if ssl can locate the information about a previous connection in the local session cache, it can reestablish the connection much more quickly than it can without the connection information.
... it is also possible for an application to import a socket into ssl after the tcp connection on that socket has already been established.
...And 3 more matches
History Service Design
history service checks for database existance and coherence, and initializes the connection through the storage service.
... database maintenance at startup the service creates an exclusive storage connection to places.sqlite, the exclusive locking is needed for both a perf gain and data-safety.
... in case the database is missing a new one is created, if instead the database exists but the connection to it fails due to database corruption, the corrupt database is moved away and a new one is created.
...And 3 more matches
Observer Notifications
profile-change-net-teardown the network connection is going offline at this point.
... http requests these are the topics that you can observe during a http request (see setting http request headers and creating sandboxed http connections).
... places-connection-closed sent after places has closed its database connection.
...And 3 more matches
mozIStorageAggregateFunction
objects implementing this interface can be registered with mozistorageconnection.createaggregatefunction().
...if you use mozistorageconnection.executeasync() or, mozistoragestatement.executeasync() this callback will run on a different thread from the rest of your code.
...if you use mozistorageconnection.executeasync() or, mozistoragestatement.executeasync() this callback will run on a different thread from the rest of your code.
...And 3 more matches
mozIStorageStatement
storage/public/mozistoragestatement.idlscriptable this interface lets you create and execute sql statements on a mozistorageconnection.
... method overview void initialize(in mozistorageconnection adbconnection, in autf8string asqlstatement); obsolete since gecko 1.9.1 void finalize(); mozistoragestatement clone(); autf8string getparametername(in unsigned long aparamindex); unsigned long getparameterindex(in autf8string aname); autf8string getcolumnname(in unsigned long acolumnindex); unsigned long getcolumnindex(in autf8string aname); void reset(); astring escapestringforlike(in astring avalue, in wchar aescapechar); void bindparameters(in mozisto...
...to initialize a statement, consumers should call mozistorageconnection.createstatement().
...And 3 more matches
nsIProtocolProxyService
otherwise, this method returns null indicating that a direct connection should be used.
... return value an nsiproxyinfo object or null for a direct connection.
...this method may be useful in conjunction with nsisockettransportservice.createtransport() for creating, for example, a socks connection.
...And 3 more matches
RTCConfiguration.iceServers - Web APIs
]; value an array of zero or more rtciceserver objects, each of which describes one stun or turn server for the ice agent to use during the connection's negotiation.
... if the array is empty, or if the iceservers option isn't specified, the ice agent will negotiate without the use of any servers, which will limit the connection to local peers.
...while it can be useful to provide a second server as a fallback in case the first is offline, listing too many servers can delay the user's connection being established, depending on the network's performance and how many servers get used for negotiation before a connection is established.
...And 3 more matches
RTCIceTransportState - Web APIs
"checking" at least one remote candidate has been received, and the rtcicetransport has begun examining pairings of remote and local candidates in order to attempt to identify viable pairs that could be used to establish a connection.
...a value of "disconnected" means that a transient issue has occurred that has broken the connection, but that should resolve itself automatically without your code having to take any action.
...this is a terminal state, indicating that the connection cannot be achieved or maintained.
...And 3 more matches
RTCRtpTransceiver.direction - Web APIs
value rtcrtpsender behavior rtcrtpreceiver behavior "sendrecv" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... "sendonly" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... exceptions when setting the value of direction, the following exceptions can occur: invalidstateerror either the receiver's rtcpeerconnection is closed or the rtcrtpreceiver is stopped.
...And 3 more matches
Proxy Auto-Configuration (PAC) file - HTTP
return value format the javascript function returns a single string if the string is null, no proxies should be used the string can contain any number of the following building blocks, separated by a semicolon: direct connections should be made directly, without any proxies proxy host:port the specified proxy should be used socks host:port the specified socks server should be used recent versions of firefox support as well: http host:port the specified proxy should be used https host:port the specified https proxy should be used socks4 host:port socks5 host:port the specified socks server (with the specifi...
...ed sock version) should be used if there are multiple semicolon-separated settings, the left-most setting will be used, until firefox fails to establish the connection to the proxy.
... if all proxies are down, and there was no direct option specified, the browser will ask if proxies should be temporarily ignored, and direct connections attempted.
...And 3 more matches
TCP/IP Security - Archive of obsolete content
this layer provides connection-oriented or connectionless services for transporting application layer services between networks.
... data link layer controls are applied to all communications on a specific physical link, such as a dedicated circuit between two buildings or a dial-up modem connection to an internet service provider (isp).
... data link layer controls for dedicated circuits are most often provided by specialized hardware devices known as data link encryptors; data link layer controls for other types of connections, such as dial-up modem communications, are usually provided through software.
...And 2 more matches
How do you upload your files to a web server? - Learn web development
note: make sure your hosting provider offers sftp (secure ftp) connection to your hosting space.
...respectively, these are a log of messages showing the connection status between your computer and the sftp server, and a live log of every interaction between your sftp client and the server.
... of course, it is a good idea to use a secure connection, like with ftp.
...And 2 more matches
Error codes returned by Mozilla APIs
network errors the following errors can occur when creating connections over the network, or when reading or writing data.
... ns_error_already_connected (0x804b000b) the connection is already established.
... ns_error_not_connected (0x804b000c) the connection does not exist.
...And 2 more matches
HTTP delegation
therefore the api should allow for keep-alive (persistent) http connections.
...the object may be used by the application to associate it with a physical network connection.
...should the application be unable to keep a physical connection alive all the time, the application is expected to create new connections automatically.
...And 2 more matches
HTTP delegation
therefore the api should allow for keep-alive (persistent) http connections.
...the object may be used by the application to associate it with a physical network connection.
...should the application be unable to keep a physical connection alive all the time, the application is expected to create new connections automatically.
...And 2 more matches
Notes on TLS - SSL 3.0 Intolerant Servers
the connection seems terminated and a blank page is displayed.
...this incorrect implementation causes them to reject connection attempts from clients that are compliant with the ssl 3.0 and tls (aka ssl 3.1) specifications.
...it is designed to permit a server to detect a man-in-the-middle that is altering the ssl client hello (connection) requests as they pass from the client to the server, altering them by changing the protocol version number to a lower version number.
...And 2 more matches
MailNews fakeserver
onstartup none server's response called when a connection is made.
... server.start(port); // set up a nsimsgincomingserver locally localserver.someactionrequiringconnection(); server.performtest(); // nothing will be executed until the connection is closed // localserver.closecachedconnections() is generally a good way to do so server.resettest(); // set up second test server.performtest(); transaction = server.playtransaction(); // finished with tests server.stop(); } currently, fakeserver provides no means to keep a persistent connection past ...
...a test, requiring connections to be closed, possibly forcibly.
...And 2 more matches
Inspecting web sockets - Firefox Developer Tools
since firefox 71, the network monitor has had the ability to inspect web socket connections.
... accessing the inspector when you are inspecting a web app that utilizes a web socket connection, the web socket requests are listed in the list of requests in the network monitor along with all other requests.
... you can use the ws button to filter the list for just web socket connections.
...And 2 more matches
about:debugging - Firefox Developer Tools
when about:debugging opens, on the left-hand side, you'll see a sidebar with two options and information about your remote debugging setup: setup use the setup tab to configure the connection to your remote device.
... to start a debugging session, first open the page that you wish to debug and then click connect next to the device name to open a connection to it.
... if the connection was successful, you can now click the name of the device to switch to a tab with information about the device.
...And 2 more matches
IDBDatabase - Web APIs
the idbdatabase interface of the indexeddb api provides a connection to a database; you can use an idbdatabase object to open a transaction on your database then create, manipulate, and delete objects (data) in that database.
... methods inherits from: eventtarget idbdatabase.close() returns immediately and closes the connection to a database in a separate thread.
... abort fired when when a transaction is aborted and bubbles up to the connection object.
...And 2 more matches
Using the Resource Timing API - Web APIs
resource loading phases an application can get timestamps for the various phases of resource loading such as redirection, dns lookup, and tcp connection setup.
...tcp time = " + t); // secure connection time t = (resources[i].secureconnectionstart > 0) ?
... (resources[i].connectend - resources[i].secureconnectionstart) : "0"; console.log("...
...And 2 more matches
WebSocket.readyState - Web APIs
the websocket.readystate read-only property returns the current state of the websocket connection.
...the connection is not yet open.
... 1 open the connection is open and ready to communicate.
...And 2 more matches
Writing a WebSocket server in Java - Web APIs
this server conforms to rfc 6455, so it only handles connections from chrome version 16, firefox 11, ie 10 and higher.
... first steps websockets communicate over a tcp (transmission control protocol) connection.
...t java.security.nosuchalgorithmexception; import java.util.base64; import java.util.scanner; import java.util.regex.matcher; import java.util.regex.pattern; public class websocket { public static void main(string[] args) throws ioexception, nosuchalgorithmexception { serversocket server = new serversocket(80); try { system.out.println("server has started on 127.0.0.1:80.\r\nwaiting for a connection..."); socket client = server.accept(); system.out.println("a client connected."); socket methods: java.net.socket getinputstream() returns an input stream for this socket.
...And 2 more matches
The WebSocket API (WebSockets) - Web APIs
note: while a websocket connection is functionally somewhat similar to standard unix-style sockets, they are not related.
... interfaces websocket the primary interface for connecting to a websocket server and then sending and receiving data on the connection.
... closeevent the event sent by the websocket object when the connection closes.
...And 2 more matches
408 Request Timeout - HTTP
WebHTTPStatus408
the hypertext transfer protocol (http) 408 request timeout response status code means that the server would like to shut down this unused connection.
... it is sent on an idle connection by some servers, even without any previous request by the client.
... a server should send the "close" connection header field in the response, since 408 implies that the server has decided to close the connection rather than continue waiting.
...And 2 more matches
HTTP response status codes - HTTP
WebHTTPStatus
408 request timeout this response is sent on an idle connection by some servers, even without any previous request by the client.
... it means that the server would like to shut down this unused connection.
... this response is used much more since some browsers, like chrome, firefox 27+, or ie9, use http pre-connection mechanisms to speed up surfing.
...And 2 more matches
Using dns-prefetch - Web Performance
for websites that open connections to many third parties, this latency can significantly reduce loading performance.
...while dns-prefetch only performs a dns lookup, preconnect establishes a connection to a server.
... this process includes dns resolution, as well as establishing the tcp connection, and performing the tls handshake—if a site is served over https.
...And 2 more matches
Transmission Control Protocol (TCP) - MDN Web Docs Glossary: Definitions of Web-related terms
tcp (transmission control protocol) is a transport layer host-to-host protocol for connection-oriented communication between two computers on an ip network.
... tcp uses virtual ports to create a virtual end-to-end connection that can reuse the physical connections between two computers.
... tcp handshake the tcp three-way handshake, also called the tcp-handshake, three message handshake, and/or syn-syn-ack, is the method used by tcp to set up a tcp/ip connection over an ip-based network.
...the three message mechanism is designed for the two computers that want to pass information back and forth and can negotiate the parameters of the connection before transmitting data such as http browser requests.
Client-Server Overview - Learn web development
the first part is called the header, and contains useful information about the request, in the same way that an html head contains useful information about an html document (but not the actual content itself, which is in the body): get https://developer.mozilla.org/search?q=client+server+overview&topic=apps&topic=html&topic=css&topic=js&topic=api&topic=webdev http/1.1 host: developer.mozilla.org connection: keep-alive pragma: no-cache cache-control: no-cache upgrade-insecure-requests: 1 user-agent: mozilla/5.0 (windows nt 10.0; wow64) applewebkit/537.36 (khtml, like gecko) chrome/52.0.2743.116 safari/537.36 accept: text/html,application/xhtml+xml,application/xml;q=0.9,image/webp,*/*;q=0.8 referer: https://developer.mozilla.org/ accept-encoding: gzip, deflate, sdch, br accept-charset: iso-8859-1,utf...
... http/1.1 200 ok server: apache x-backend-server: developer1.webapp.scl3.mozilla.com vary: accept,cookie, accept-encoding content-type: text/html; charset=utf-8 date: wed, 07 sep 2016 00:11:31 gmt keep-alive: timeout=5, max=999 connection: keep-alive x-frame-options: deny allow: get x-cache-info: caching content-length: 41823 <!doctype html> <html lang="en-us" dir="ltr" class="redesign no-js" data-ffo-opensanslight=false data-ffo-opensans=false > <head prefix="og: http://ogp.me/ns#"> <meta charset="utf-8"> <meta http-equiv="x-ua-compatible" content="ie=edge"> <script>(function(d) { d.classname = d.classname.replace(/\bno...
... post https://developer.mozilla.org/profiles/hamishwillee/edit http/1.1 host: developer.mozilla.org connection: keep-alive content-length: 432 pragma: no-cache cache-control: no-cache origin: https://developer.mozilla.org upgrade-insecure-requests: 1 user-agent: mozilla/5.0 (windows nt 10.0; wow64) applewebkit/537.36 (khtml, like gecko) chrome/52.0.2743.116 safari/537.36 content-type: application/x-www-form-urlencoded accept: text/html,application/xhtml+xml,application/xml;q=0.9,image/webp,*/*;q=0.8 refer...
... http/1.1 302 found server: apache x-backend-server: developer3.webapp.scl3.mozilla.com vary: cookie vary: accept-encoding content-type: text/html; charset=utf-8 date: wed, 07 sep 2016 00:38:13 gmt location: https://developer.mozilla.org/profiles/hamishwillee keep-alive: timeout=5, max=1000 connection: keep-alive x-frame-options: deny x-cache-info: not cacheable; request wasn't a get or head content-length: 0 note: the http responses and requests shown in these examples were captured using the fiddler application, but you can get similar information using web sniffers (e.g.
JSS FAQ
MozillaProjectsNSSJSSJSS FAQ
my ssl connection is hanging on windows?
... how can i debug my ssl connection?
... my ssl connection is hanging on windows?
... check http://lxr.mozilla.org/mozilla/source/security/jss/org/mozilla/jss/ssl/sslsocket.java#730 how can i debug my ssl connection?
nsIBadCertListener2
it can also be used to obtain the ssl handshake status of a connection that will be canceled because of improper certificate status.
...boolean notifycertproblem( in nsiinterfacerequestor socketinfo, in nsisslstatus status, in autf8string targetsite ); parameters socketinfo a network communication context that can be used to obtain more information about the active connection.
...targetsite the site name that was used to open the current connection.
... return value the consumer shall return true if it wants to suppress the error message related to the bad cert (the connection will still get canceled).
nsISSLErrorListener
security/manager/ssl/public/nsisslerrorlistener.idlscriptable a mechanism to report a broken ssl connection.
...boolean notifysslerror( in nsiinterfacerequestor socketinfo, in print32 error, in autf8string targetsite ); parameters socketinfo a network communication context that can be used to obtain more information about the active connection.
...targetsite the site name that was used to open the current connection.
... return value the consumer shall return true if it wants to suppress the error message related to the error (the connection will still get canceled).
nsITransportSecurityInfo
netwerk/socket/nsitransportsecurityinfo.idlscriptable this interface provides information about transport security, including the security state and any error message for the connection.
... inherits from: nsisupports last changed in gecko 1.9 (firefox 3) attributes attribute type description errormessage wstring error message on connection failure.
... securitystate unsigned long a flag detailing the security state of the connection.
... shortsecuritydescription wstring for secure connections (ssl) gives the common name (cn) of the certifying authority.
IDBDatabase: close event - Web APIs
the close event is fired on idbdatabase when the database connection is unexpectedly closed.
... note that it is not fired if the database connection is closed normally using idbdatabase.close().
...; objectstore.createindex('minutes', 'minutes', { unique: false }); objectstore.createindex('day', 'day', { unique: false }); objectstore.createindex('month', 'month', { unique: false }); objectstore.createindex('year', 'year', { unique: false }); }; dbopenrequest.onsuccess = (event) => { const db = dbopenrequest.result; db.addeventlistener('close', () => { console.log('database connection closed'); }); }; the same example, using the onclose property instead of addeventlistener(): // open the database const dbopenrequest = window.indexeddb.open('todolist', 4); dbopenrequest.onupgradeneeded = (event) => { const db = event.target.result; // create an objectstore for this database const objectstore = db.createobjectstore('todolist', { keypath: 'tasktitle' }); // defin...
...unique: false }); objectstore.createindex('minutes', 'minutes', { unique: false }); objectstore.createindex('day', 'day', { unique: false }); objectstore.createindex('month', 'month', { unique: false }); objectstore.createindex('year', 'year', { unique: false }); }; dbopenrequest.onsuccess = (event) => { const db = dbopenrequest.result; db.onclose = () => { console.log('database connection closed'); }; }; ...
IDBFactory.open() - Web APIs
WebAPIIDBFactoryopen
the open() method of the idbfactory interface requests opening a connection to a database.
...if the operation is successful, a success event is fired on the request object that is returned from this method, with its result attribute set to the new idbdatabase object for the connection.
... if an error occurs while the database connection is being opened, then an error event is fired on the request object returned from this method.
...if the version is not provided and the database exists, then a connection to the database will be opened without changing its version.
Basic concepts - Web APIs
database connection an operation created by opening a database.
... a given database can have multiple connections at the same time.
... a database connection can have several active transactions associated with it at a time, so long as the writing transactions do not have overlapping scopes.
...so, for example, if a database connection already has a writing transaction with a scope that just covers the flyingmonkey object store, you can start a second transaction with a scope of the unicorncentaur and unicornpegasus object stores.
NetworkInformation.downlinkMax - Web APIs
the networkinformation.downlinkmax read-only property returns the maximum downlink speed, in megabits per second (mbps), for the underlying connection technology.
... syntax var max = networkinformation.downlinkmax return value an unrestricted double representing the maximum downlink speed, in megabits per second (mb/s), for the underlying connection technology.
... examples the following example monitors the connection using the change event and logs changes as they occur.
... function logconnectiontype() { var connectiontype = 'not supported'; var downlinkmax = 'not supported'; if ('connection' in navigator) { connectiontype = navigator.connection.effectivetype; if ('downlinkmax' in navigator.connection) { downlinkmax = navigator.connection.downlinkmax; } } console.log('current connection type: ' + connectiontype + ' (downlink max: ' + downlinkmax + ')'); } logconnectiontype(); navigator.connection.addeventlistener('change', logconnectiontype); specifications specification status comment network information apithe definition of 'downlinkmax' in that specification.
NetworkInformation.onchange - Web APIs
the networkinformation.onchange event handler contains the code that is fired when connection information changes, and the change is received by the networkinformation object.
...} examples // get the connection type.
... var type = navigator.connection.type; function changehandler(e) { // handle change of connection type here.
... } // register for event changes: navigator.connection.onchange = changehandler; // another way: navigator.connection.addeventlistener('change', changehandler); specifications specification status comment network information apithe definition of 'onchange' in that specification.
NetworkInformation.rtt - Web APIs
the networkinformation.rtt read-only property returns the estimated effective round-trip time of the current connection, rounded to the nearest multiple of 25 milliseconds.
... this value is based on recently observed application-layer rtt measurements across recently active connections.
... it excludes connections made to a private address space.
... if no recent measurement data is available, the value is based on the properties of the underlying connection technology.
PerformanceResourceTiming.connectEnd - Web APIs
the connectend read-only property returns the timestamp immediately after the browser finishes establishing the connection to the server to retrieve the resource.
... the timestamp value includes the time interval to establish the transport connection, as well as other time intervals such as ssl handshake and socks authentication.
... syntax resource.connectend; return value a domhighrestimestamp representing the time after a connection is established.
...; } } function print_start_and_end_properties(perfentry) { // print timestamps of the *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming - Web APIs
performanceresourcetiming.connectstartread only a domhighrestimestamp immediately before the browser starts to establish the connection to the server to retrieve the resource.
... performanceresourcetiming.connectendread only a domhighrestimestamp immediately after the browser finishes establishing the connection to the server to retrieve the resource.
... performanceresourcetiming.secureconnectionstartread only a domhighrestimestamp immediately before the browser starts the handshake process to secure the current connection.
... performanceresourcetiming.responseendread only a domhighrestimestamp immediately after the browser receives the last byte of the resource or immediately before the transport connection is closed, whichever comes first.
PerformanceTiming.connectEnd - Web APIs
the legacy performancetiming.connectend read-only property returns an unsigned long long representing the moment, in miliseconds since the unix epoch, where the connection is opened network.
... if the transport layer reports an error and the connection establishment is started again, the last connection establisment end time is given.
... if a persistent connection is used, the value will be the same as performancetiming.fetchstart.
... a connection is considered as opened when all secure connection handshake, or socks authentication, is terminated.
RTCDataChannel.label - Web APIs
a data channel's label is set when the channel is created by calling rtcpeerconnection.createdatachannel().
... syntax var name = adatachannel.label; value a string identifier assigned by the web site or app when the data channel was created, as specified when rtcpeerconnection.createdatachannel() was called to create the channel.
... example this sample creates a data channel on an rtcpeerconnection, then, some time later, sets the content of a ui element to display the channel's name.
... var pc = new rtcpeerconnection(); var dc = pc.createdatachannel("my channel"); /* ...
RTCDataChannel.send() - Web APIs
data sent before connecting is buffered if possible (or an error occurs if it's not possible), and is also buffered if sent while the connection is closing or closed.
... syntax rtcdatachannel.send(data); parameters data the data to transmit across the connection.
... exceptions invalidstateerror since the data channel uses a separate transport channel from the media content, it must establish its own connection; if it hasn't finished doing so (that is, its readystate is "connecting"), this error occurs without sending or buffering the data.
... var pc = new rtcpeerconnection(); var dc = pc.createdatachannel("backchannel"); function sendmessage(msg) { let obj = { "message": msg, "timestamp": new date() } dc.send(json.stringify(obj)); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.send()' in that specification.
RTCIceCandidate.usernameFragment - Web APIs
to do so, you can compare the value of usernamefragment to the current usernamefragment being used for the connection after receiving the candidate from the signaling server and before caling addicecandidate() to add it to the set of possible candidates.
... when the web app receives a message from the signaling server that includes a candidate to be added to the rtcpeerconnection, you can (and generally should) simply call addicecandidate().
...the function below, ssnewcandidate(), is called when a message, signalmsg, arrives from the signaling server that contains an ice candidate to be added to the rtcpeerconnection.
...otherwise, after checking every receiver, it adds the new candidate to the connection.
RTCIceCandidatePairStats.circuitBreakerTriggerCount - Web APIs
the rtcicecandidatepairstats property circuitbreakertriggercount indicates the number of times the circuit-breaker has been triggered to indicate a connection timeout or other unexpected connection abort on this specific connection configuration.
... a circuit breaker trigger is fired each time the connection times out or otherwise needs to be halted automatically.
... syntax cbtcount = rtcicecandidatepairstats.circuitbreakertriggercount; value an integer value indicating the number of times the circuit-breaker has been triggered for the 5-tuple used by this connection.
... a 5-tuple defining a tcp connection is made up of the following data: the source ip address.
RTCIceTransport: selectedcandidatepairchange event - Web APIs
a selectedcandidatepairchange event is sent to an rtcicetransport when the ice agent selects a new pair of candidates that describe the endpoints of a viable connection.
... the pair of candidates is in turn described by an rtcicecandidatepair object which contains one rtcicecandidate representing the local end of the connection, and another representing the remote end of the connection.
... together, the candidates can be used to establish a connection to be used by the rtcicetransport, and, by extension, by an rtcpeerconnection.
... bubbles no cancelable no interface event event handler property onselectedcandidatepairchange examples this example creates an event handler for selectedcandidatepairchange that updates a display providing the user information about the progress of the ice negotiation for an rtcpeerconnection called pc.
RTCRemoteOutboundRtpStreamStats.localId - Web APIs
networkteststart() this function simply calls the rtcpeerconnection method getstats() to request an rtcstatsreport and store it in the variable startreport.
... let startreport; async function networkteststart(pc) { if (pc) { startreport = await pc.getstats(); } } given an rtcpeerconnection, pc, this calls its getstats() method to obtain a statistics report object, which it stores in startreport for use once the end-of-test data has been collected by networkteststop().
...it takes as input the rtcpeerconnection being tested, calls getstats() to get a new rtcstatsreport with current statistics, then computes the results it's looking for, outputting those results as appropriate to the user by appending appropriate html to the contents of the <div> element whose class is stats-box.
... `data size: ${bytessent} bytes.</div>`; statsbox.innerhtml += logentry; } else { statsbox.innerhtml += `<div class="stats-error">unable to find initial statistics for id ${endremoteoutbound.id}.</div>` } } statsbox.scrollto(0, statsbox.scrollheight); } } } here's what's going on in the networkteststop() function: after calling the rtcpeerconnection method getstats() to get the latest statistics report for the connection and storing it in endreport, this is an rtcstatsreport object, which maps strings taken from the rtcstatstype enumerated type to objects of the corresponding rtcstats-based type.
RTCRtpSender.setStreams() - Web APIs
exceptions invalidstateerror the sender's connection is closed.
... once the track has been added to all of the streams, renegotiation of the connection will be triggered by the negotiationneeded event being dispatched to the rtcpeerconnection to which the sender belongs.
... example this example adds all of an rtcpeerconnection's tracks to the specified stream.
... function addtrackstostream(stream) { let senders = pc.getsenders(); senders.foreach((sender) => { if (sender.track && (sender.transport.state === connected)) { sender.setstreams(stream); } }); } after calling the rtcpeerconnection method getsenders() to get the list of the connection's senders, the addtrackstostream() function iterates over the list.
RTCRtpTransceiver.stop() - Web APIs
note: until recently, the stopped property was provided to return true if the connection is stopped.
... return value undefined exceptions invalidstateerror the rtcpeerconnection of which the transceiver is a member is closed.
...the receiver then stops receiving media; the receiver's track is stopped, and the transceiver's direction is changed to stopped, and renegotiation is triggered by sending a negotiationneeded event to the rtcpeerconnection.
... note: stopping the transceiver causes a negotiationneeded event to be sent to the transceiver's rtcpeerconnection, so the connection can adapt to the change.
WebSocket.close() - Web APIs
WebAPIWebSocketclose
the websocket.close() method closes the websocket connection or connection attempt, if any.
... if the connection is already closed, this method does nothing.
... syntax websocket.close(); parameters code optional a numeric value indicating the status code explaining why the connection is being closed.
... reason optional a human-readable string explaining why the connection is closing.
Using Web Workers - Web APIs
the port connection needs to be started either implicitly by use of the onmessage event handler or explicitly with the start() method before any messages can be posted.
... note: when using the start() method to open the port connection, it needs to be called from both the parent thread and the worker thread if two-way communication is needed.
...there is a bit more complexity here as well (worker.js): onconnect = function(e) { var port = e.ports[0]; port.onmessage = function(e) { var workerresult = 'result: ' + (e.data[0] * e.data[1]); port.postmessage(workerresult); } } first, we use an onconnect handler to fire code when a connection to the port happens (i.e.
...setting up this message handler in the worker thread also implicitly opens the port connection back to the parent thread, so the call to port.start() is not actually needed, as noted above.
Index - HTTP
WebHTTPHeadersIndex
21 connection http, headers, reference, web the connection general header controls whether or not the network connection stays open after the current transaction finishes.
... if the value sent is keep-alive, the connection is persistent and not closed, allowing for subsequent requests to the same server to be done.
... 82 index http, http header, index found 115 pages: 83 keep-alive general header, http, http header, reference the keep-alive general header allows the sender to hint about how the connection may be used to set a timeout and a maximum amount of requests.
...that is, this is header is sent from server to client to inform that server is willing to initiate a websocket connection.
Strict-Transport-Security - HTTP
description if a website accepts a connection through http and redirects to https, visitors may initially communicate with the non-encrypted version of the site before being redirected, if, for example, the visitor types http://www.foo.com/ or even just foo.com.
... note: the strict-transport-security header is ignored by the browser when your site is accessed using http; this is because an attacker may intercept http connections and inject the header or remove it.
...should it be necessary to disable strict transport security, setting the max-age to 0 (over a https connection) will immediately expire the strict-transport-security header, allowing access via http.
...by following the guidelines and successfully submitting your domain, browsers will never connect to your domain using an insecure connection.
HTTP Messages - HTTP
WebHTTPMessages
in http/1.1, and earlier versions of the protocol, these messages were openly sent across the connection.
... headers are often very similar from one message to the next one, yet still repeated across connections.
...several connections need opening on the same server: and warm tcp connections are more efficient than cold ones.
...several streams can be combined together, a process called multiplexing, allowing more efficient underlying tcp connections.
Transport Layer Security - Web security
the security of any connection using transport layer security (tls) is heavily dependent upon the cipher suites and security parameters selected.
... a tls connection starts with a handshake phase where a client and server agree on a shared secret and important parameters, like cipher suites, are negotiated.
... tls 1.3 supports forward-secure modes only, unless the connection is resumed or it uses a pre-shared key.
... from version 74 onwards, firefox will return a secure connection failed error when connecting to servers using the older tls versions (bug 1606734).
MCD, Mission Control Desktop, AKA AutoConfig - Archive of obsolete content
ker_mode", "0"); lockpref("mail.identity.id1.organization", "int evry france"); lockpref("mail.identity.id1.overrideglobal_pref", true); lockpref("mail.identity.id1.reply_to", ""); //imap lockpref("mail.server.server2.hostname", "imap-int.int-evry.fr"); lockpref("mail.server.server2.issecure", true); lockpref("mail.server.server2.login_at_startup", true); lockpref("mail.server.server2.max_cached_connections", 5); //lockpref("mail.server.server2.name", "jehan.procaccia@int-evry.fr"); lockpref("mail.server.server2.type", "imap"); lockpref("mail.server.server2.username", env_user); //smtp lockpref("mail.identity.id1.smtpserver", "smtp1"); lockpref("mail.identity.id1.stationery_folder", "imap://" + env_user + "@imap-int.int-evry.fr/templates"); lockpref("mail.identity.id1.tmpl_folder_picker_mode", "0"...
...k-1.mab"); lockpref("ldap_2.servers.ldapint.position", 3); lockpref("ldap_2.servers.ldapint.uri", "ldap://ldap1.int-evry.fr:389/ou=people,dc=int-evry,dc=fr??sub"); lockpref("ldap_2.servers.pab.filename", "abook.mab"); lockpref("ldap_2.servers.pab.replication.lastchangenumber", 0); //news config lockpref("mail.server.server3.hostname", "news.int-evry.fr"); lockpref("mail.server.server3.max_cached_connections", 2); lockpref("mail.server.server3.name", "news.int-evry.fr"); lockpref("mail.server.server3.type", "nntp"); lockpref("mail.server.server3.username", env_user); //call to ldap to get user's attribute.
...k-1.mab"); lockpref("ldap_2.servers.ldapint.position", 3); lockpref("ldap_2.servers.ldapint.uri", "ldap://ldap1.int-evry.fr:389/ou=people,dc=int-evry,dc=fr??sub"); lockpref("ldap_2.servers.pab.filename", "abook.mab"); lockpref("ldap_2.servers.pab.replication.lastchangenumber", 0); //news config lockpref("mail.server.server3.hostname", "news.int-evry.fr"); lockpref("mail.server.server3.max_cached_connections", 2); lockpref("mail.server.server3.name", "news.int-evry.fr"); lockpref("mail.server.server3.type", "nntp"); lockpref("mail.server.server3.username", env_user); // close the try, and call the catch() } catch(e) { displayerror("lockedpref", e); } for the record, old reliably scripts...
No Proxy For configuration - Archive of obsolete content
because the public network was small in scope and connections were slow, a caching proxy could often improve the overall performance.
...proxy connections that fail return an error "the proxy server you have configured cannot be found", so configure your browser to use a non-existent http proxy (hostname: "imaginary", port "80").
...all proxied urls will return errors, all non-proxied connections will be attempted normally (direct connection).
Remotely debugging Firefox for Metro - Archive of obsolete content
p firefox for metro go to about:config in firefox for metro, and set the following required preference: devtools.debugger.remote-enabled = true you may also want to set these optional preferences: devtools.debugger.force-local = false (if you want to connect from a different machine over the network) devtools.debugger.remote-host (to change the tcp hostname where firefox will listen for connections) devtools.debugger.remote-port (to change the tcp port number where firefox will listen for connections) devtools.debugger.prompt-connection = false (allow connections without displaying a confirmation prompt.
... if you are connecting to firefox for metro on a different computer or a different port, enter the appropriate hostname and port number and then press "connect." in the windows 8 (metro) browser next you'll see a dialog in firefox for metro asking you to confirm the connection.
... tip: if the remote connection times out, flip to the desktop via the desktop tile and check if windows firewall has thrown up an incoming connection permissions prompt.
Back to the Server: Server-Side JavaScript On The Rise - Archive of obsolete content
listing 1 - querying a mysql database from rhino // import the java sql packages importpackage( java.sql ); // load the mysql driver java.lang.class.forname( "com.mysql.jdbc.driver" ); // create connection to the database var conn = drivermanager.getconnection( "jdbc:mysql://localhost/rhino", "urhino", "prhino" ); // create a statement handle var stmt = conn.createstatement(); // get a resultset var rs = stmt.executequery( "select * from employee" ); // get the metadata from the resultset var meta = rs.getmetadata(); // loop over the records, dump out column names and values while( rs.next() ) ...
...the appropriate database driver for mysql is loaded and the connection string to a database named rhino on my local machine is configured using the user account urhino with the password of prhino.
... importpackage( java.net ); // connect to the remote resource var u = new url( "http://www.mozilla.org/news.rdf" ); var c = u.openconnection(); c.connect(); // read in the raw data var s = new java.io.inputstreamreader( c.getinputstream() ); var b = new java.io.bufferedreader( s ); var l, str = ""; while( ( l = b.readline() ) != null ) { // skip if( l != "" ) { str = str + l + "\n"; } } // define the namespaces, first the default, // then additional namespaces default xml namespace = "http://purl.org/rss/1.0/"; var dc = new name...
Domain sharding - MDN Web Docs Glossary: Definitions of Web-related terms
browsers limit the number of active connections for each domain.
... the problem with domain sharding, in terms of performance, is the cost of extra dns lookups for each domain and the overhead of establishing each tcp connection.
...as browsers limit the number of active connections per domain, serving all the required resources from a single domain could be slow as assets need to be downloaded sequentially.
QUIC - MDN Web Docs Glossary: Definitions of Web-related terms
quick udp internet connection, or quic, is an experimental multiplexed transport protocol implemented on udp.
... key features of quic include: reduction in connection establishment time.
... connection migration.
TCP slow start - MDN Web Docs Glossary: Definitions of Web-related terms
tcp slow start is an algorithm used to detect the available bandwidth for packet transmission, and balances the speed of a network connection.
...when a new connection is made, cwnd is initialized to one tcp data or acknowledgment packet, and waits for an acknowledgement, or ack.
...the connection has a limited capacity depending on hardware and network conditions.
Tips for authoring fast-loading HTML pages - Learn web development
an optimized web page not only provides for a more responsive site for your visitors but also reduces the load on your web servers and internet connection.
...it is just as important for broadband content and can lead to dramatic improvements even for your visitors with the fastest connections.
... minimize the number of files reducing the number of files referenced in a web page lowers the number of http connections required to download a page, thereby reducing the time for these requests to be sent, and for their responses to be received.
Client-side storage - Learn web development
saving data and assets locally so a site will be quicker (and potentially less expensive) to download, or be usable without a network connection.
...this api is designed for storing http responses to specific requests, and is very useful for doing things like storing website assets offline so the site can subsequently be used without a network connection.
...this is already a great improvement to the user experience, but there is still one thing missing — the main html, css, and javascript files still need to be downloaded each time the site is accessed, meaning that it won't work when there is no network connection.
Aprender y obtener ayuda - Learn web development
you let your mind wander more widely, and seemingly make random connections between different things.
...this is the opposite of focus — you let your brain wander around the wider landscape, searching around for connections you didn't have before, touching on new things (or new combinations of things) that you can then focus on later, to strengthen them and start to really understand what they mean.
...but then after going on a walk to get some fresh air, you may well find that as your mind wanders, you suddenly make a connection between tool a and tool b, and realize that you can use them together to fix problem c!
The "why" of web performance - Learn web development
with performance as with accessibility, you consider what device a site visitor is using to access the site and the device connection speed.
...a lot of countries still have internet connections that bill per megabyte.
... here's some real-world examples of performance improvements: tokopedia reduced render time from 14s to 2s for 3g connections and saw a 19% increase in visitors, 35% increase in total sessions, 7% increase in new users, 17% increase in active users and 16% increase in sessions per user.
Storage access policy: Block cookies from trackers
network connections: tls sessions will not be resumed using a session ticket when an https connection is made to an embedded third-party resource that is classified as a tracker.
... http connection reuse by domains classified as trackers is limited to requests that occur under the same top-level origin.
... for example, a request for content from tracker.example on news.example will not reuse an http connection with a request for content from tracker.example on shopping.example or with requests that occur when tracker.example is visited directly (i.e., as a first party).
Release phase
let's go over configuring your connection protocols.
...connection to hg.mozilla.org closed.
... if you don't see this message, try running the same command in a more verbose mode (with the debugging information): $ ssh -vvv hg.mozilla.org this should tell you why your connection is not succeeding.
NSPR Error Handling
pr_connect_refused_error the peer has refused to allow the connection to be established.
... pr_connect_timeout_error the connection attempt did not complete in a reasonable period of time.
... pr_connect_reset_error the (tcp) connection has been reset by the peer.
PRIOMethods
accept accept a connection from a network peer.
... listen prepare to listen for network connections.
... shutdown shut down a network connection.
PR_AcceptRead
accepts a new connection and receives a block of data.
...pr_acceptread blocks indefinitely until the connection is accepted; the read will time out after the timeout interval elapses.
... description pr_acceptread accepts a new connection and retrieves the newly created socket's descriptor and the connecting peer's address.
PR_GetConnectStatus
get the completion status of a nonblocking connection.
... if pr_geterror returns pr_in_progress_error, the nonblocking connection is still in progress and has not completed yet.other errors indicate that the connection has failed.
... description after pr_connect on a nonblocking socket fails with pr_in_progress_error, you may wait for the connection to complete by calling pr_poll on the socket with the in_flags pr_poll_write | pr_poll_except.
NSS 3.12.6 release notes
new in nss 3.12.6 ssl3 & tls renegotiation indication extension (rfc 5746) by default, nss 3.12.6 uses the new tls renegotiation indication extension for tls renegotiation but allows simple ssl/tls connections (without renegotiation) with peers that don't support the tls renegotiation indication extension.
...if true, a connection will be dropped at initial handshake if the peer server or client does not support safe renegotiation.
...this allows packet sniffers to decrypt tls connections.
Python binding for NSS
-10-29 scm tag pynss_release_0_16_0 source download https://ftp.mozilla.org/pub/mozilla.org/security/python-nss/releases/pynss_release_0_16_0/src/ change log the primary enhancements in this version is adding support for the setting trust attributes on a certificate, the ssl version range api, information on the ssl cipher suites and information on the ssl connection.
... the following classes were added: sslciphersuiteinfo sslchannelinfo the following class methods were added: certificate.trust_flags certificate.set_trust_attributes sslsocket.set_ssl_version_range sslsocket.get_ssl_version_range sslsocket.get_ssl_channel_info sslsocket.get_negotiated_host sslsocket.connection_info_format_lines sslsocket.connection_info_format sslsocket.connection_info_str sslciphersuiteinfo.format_lines sslciphersuiteinfo.format sslchannelinfo.format_lines sslchannelinfo.format the following class properties were added: certificate.ssl_trust_flags certificate.email_trust_flags certificate.signing_t...
...view all ref counting, numerous ref counting fixes implement cyclic garbage collection support by adding object traversal and clear methods identify static variables, move to thread local storage remove python-nss specific httplib.py, no longer needed python-nss now compatible with standard library rewrite httplib_example.py to use standard library and illustrate ssl, non-ssl, connection class, http class usage the following classes were added: authkeyid basicconstraints crldistributionpoint crldistributionpts certificateextension generalname signedcrl dn rdn ava certificaterequest the following module functions were added: nss.nss.nss_is_initialized() nss.nss.cert_crl_reason_from_nam...
troubleshoot.html
if the build fails early on the gmakein coreconf try updating your cvs tree with -p: cd mozilla cvs update -p building a 32-bit version on a 64-bit may fail with: /usr/include/features.h:324:26: fatal error: bits/predefs.h: no such file or directory in this case remember to set use_64=1 testing nss the ssl stress test opens 2,048 tcp connections in quick succession.
... kernel data structures may remain allocated for these connections for up to two minutes.
... some systems may not be configured to allow this many simultaneous connections by default; if the stress tests fail, try increasing the number of simultaneous sockets supported.
sslerr.html
ssl_error_close_notify_alert -12230 "ssl peer has closed this connection." the local socket received an ssl3 alert record from the remote peer, reporting that the remote peer has chosen to end the connection.
...such a connection cannot be permitted without violating u.s.
...likely causes include that the peer has closed the connection.
mozIStorageProgressHandler
last changed in gecko 1.8 (firefox 1.5 / thunderbird 1.5 / seamonkey 1.0) inherits from: nsisupports method overview boolean onprogress(in mozistorageconnection aconnection); methods onprogress() the onprogress() method is called periodically while an sqlite operation is ongoing.
... boolean onprogress( in mozistorageconnection aconnection ); parameters <tt>aconnection</tt> a mozistorageconnection connection indicating the connection for which the callback was invoked.
... see also storage mozstorage introduction and how-to article mozistorageconnection database connection to a specific file or in-memory data storage mozistoragestatement create and execute sql statements on a sqlite database.
nsIMsgProtocolInfo
methods getdefaultserverport() returns the default port for a connection to a server of this account type.
... long getdefaultserverport( in boolean issecure ); parameters issecure whether or not the connection will be secure.
... return value the default port number for connections of this account type.
nsISSLSocketControl
netwerk/socket/nsisslsocketcontrol.idlscriptable this interface establishes tls and ssl connections.
... inherits from: nsisupports last changed in gecko 1.9 (firefox 3) method overview void proxystartssl(); void starttls(); attributes attribute type description forcehandshake boolean obsolete since gecko 1.9 notificationcallbacks nsiinterfacerequestor methods proxystartssl() starts an ssl proxy connection.
...starttls() establishes a starttls connection.
nsIWebSocketListener
no additional messages through onmessageavailable(), onbinarymessageavailable()() or onacknowledge()() will be delievered to the listener after onserverclose() is called, but outgoing messages can still be sent through the nsiwebsocketchannel connection.
... areason a text string explaining the reason for the closure of the connection.
...this is the final notification the listener will receive; once it's been received, the websocket connection is complete.
Performance
in order to keep the cache active between transactions, you needed to have a second connection with a 'dummy' (and schema changing) transaction.
...sharding the cache between connections by default, storage enables the sqlite shared-cache mode which makes multiple connections to the same database share the same cache.
...unfortunately, this operation cannot be controlled on a per-transaction or per-connection basis.
Using the Multiple Accounts API
the only thing that determines which smtp server a user should use is their physical connection to the internet.
... preference: mail.server.server.max_cached_connections - integer, max number of connections left open to the server preference: mail.server.server.empty_trash_threshhold integer, (should not be imap-specific) max k of wasted diskspace before we purge a folder preference: mail.server.server.delete_model - integer, delete model (move to trash, imap delete, purge immediately, not sure of values) preference: mail.server.server.timeout...
... - integer, number of minutes a connection is idle before we drop it preference: mail.server.server.capability - list of capabilities of this server preference: mail.server.server.namespace.public - the server's namespace for public folders preference: mail.server.server.namespace.personal - the server's namespace for personal folders preference: mail.server.server.namespace.other_users - the server's namespace for other user's folders the following are specific to pop: the following are specific to news: preference: mail.server.server.leave_on_server - boolean, should we leave messages on the server after we have downloaded them?
WebIDL bindings
for example: var contentobject = new contentwin.rtcpeerconnection(); the returned object will be an xray wrapper for the content-side object.
...for example: var contentobject = rtcpeerconnection._create(contentwin, new mypeerconnectionimpl()); however, if you are in a js component, you may only be able to get to the correct interface object via some window object.
... in this case, the code would look more like: var contentobject = contentwin.rtcpeerconnection._create(contentwin, new mypeerconnectionimpl()); creating the object this way will not invoke its __init method or init method.
Mozilla
creating sandboxed http connections starting with gecko 1.8.1 (firefox 2), it is possible to create sandboxed http connections which don't affect the user's cookies.
... this article will cover the basics of doing http connections from xpcom javascript, and should easily translate to c++ xpcom.
... mozilla web developer faq this document answers questions that web authors ask frequently specifically in connection with firefox and other gecko-based browsers.
Web Audio Editor - Firefox Developer Tools
connections to audioparams displaying connections to audioparams is new in firefox 34.
... connections between nodes are displayed as solid lines.
... if, instead, you've connected a node to an audioparam in another node, then the connection is shown as a dashed line between the nodes, and is labeled with the name of the audioparam: inspecting and modifying audionodes if you click on a node, it's highlighted and you get a node inspector on the right hand side.
AudioNode.disconnect() - Web APIs
if no parameters are provided, all outgoing connections are disconnected.
...if this value is an audioparam, then the connection to that audioparam is terminated, and the node's contributions to that computed parameter become 0 going forward once the change takes effect.
... if no matching connection is found, an invalidaccesserror exception is thrown.
CloseEvent() - Web APIs
closeeventinit optional is a closeeventinit dictionary, having the following fields: "wasclean", optional and defaulting to false, of type long, indicates if the connection has been closed cleanly or not.
... "code", optional and defaulting to 0, of type unsigned short, that is the connection close code sent by the server.
... "reason", optional and defaulting to '', of type domstring, that is a human-readable reason why the server closed the connection.
EventSource.close() - Web APIs
WebAPIEventSourceclose
the close() method of the eventsource interface closes the connection, if one is made, and sets the eventsource.readystate attribute to 2 (closed).
... note: if the connection is already closed, the method does nothing.
... examples var button = document.queryselector('button'); var evtsource = new eventsource('sse.php'); button.onclick = function() { console.log('connection closed'); evtsource.close(); } note: you can find a full example on github — see simple sse demo using php.
IDBDatabase.close() - Web APIs
WebAPIIDBDatabaseclose
the close() method of the idbdatabase interface returns immediately and closes the connection in a separate thread.
... the connection is not actually closed until all transactions created using this connection are complete.
... no new transactions can be created for this connection once this method is called.
IDBTransaction.db - Web APIs
WebAPIIDBTransactiondb
the db read-only property of the idbtransaction interface returns the database connection with which this transaction is associated.
...at the end, we return the associated database connection using db.
...ewitem object to the object store var objectstorerequest = objectstore.add(newitem[0]); objectstorerequest.onsuccess = function(event) { // report the success of the request (this does not mean the item // has been stored successfully in the db - for that you need transaction.onsuccess) note.innerhtml += '<li>request successful.</li>'; }; // return the database (idbdatabase) connection with which this transaction is associated transaction.db; }; specification specification status comment indexed database api 2.0the definition of 'db' in that specification.
PerformanceResourceTiming.connectStart - Web APIs
the connectstart read-only property returns the timestamp immediately before the user agent starts establishing the connection to the server to retrieve the resource.
... syntax resource.connectstart; return value a domhighrestimestamp immediately before the browser starts to establish the connection to the server to retrieve the resource.
...print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.responseEnd - Web APIs
the responseend read-only property returns a timestamp immediately after the browser receives the last byte of the resource or immediately before the transport connection is closed, whichever comes first.
... syntax resource.responseend; return value a domhighrestimestamp immediately after the browser receives the last byte of the resource or immediately before the transport connection is closed, whichever comes first.
...print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceTiming.connectStart - Web APIs
the legacy performancetiming.connectstart read-only property returns an unsigned long long representing the moment, in miliseconds since the unix epoch, where the request to open a connection is sent to the network.
... if the transport layer reports an error and the connection establishment is started again, the last connection establisment start time is given.
... if a persistent connection is used, the value will be the same as performancetiming.fetchstart.
RTCConfiguration.iceTransportPolicy - Web APIs
the webrtc device api dictionary rtcconfiguration's icetransportpolicy property is a string indicating the transport selection policy the ice agent should use during negotiation of connections.
... examples in this example, a new connection is configured to only accept relay candidates.
... let config = { iceservers: [ { urls: [ "stun:stun.example.com" ] }, ], icetransportpolicy: "relay" }; let pc = new rtcpeerconnection(config); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtccandidate.icetransportpolicy' in that specification.
RTCDataChannel.negotiated - Web APIs
the read-only rtcdatachannel property negotiated indicates whether the rtcdatachannel's connection was negotiated by the web app (true) or by the webrtc layer (false).
... syntax var negotiated = adatachannel.negotiated; value true if the rtcdatachannel's connection was negotiated by the web app itself; false if the negotiation was handled by the webrtc layer.
... example the code snippet below checks the value of negotiated; if it's true, a function called shutdownremotechannel() is called with the channel's id; presumably this would be implemented to transmit a shutdown signal to the remote peer prior to terminating the connection.
RTCDataChannelEvent - Web APIs
these events sent to an rtcpeerconnection when its remote peer is asking to open an rtcdatachannel between the two peers.
...just listen for the datachannel event to be received by the rtcpeerconnection and when you receive it, use the rtcdatachannelevent.channel property to gain access to the data channel which has been opened.
...the channel property provides the rtcdatachannel representing the connection to the other peer.
RTCIceCandidate.candidate - Web APIs
function handlenewicecandidate(candidatesdp) { var candidateobj = new rtcicecandidate(candidatesdp); mypeerconnection.addicecandidate(candidateobj).catch({ /* handle the error thrown by addicecandidate() */ }); } the handlenewicecandidate() function shown here passes the received candidate's sdp text into rtcicecandidate() to receive an rtcicecanddiate object in return, which represents the candidate.
... the new candidate is then passed into rtcpeerconnection.addicecandidate() to add the candidate to the list of candidates for webrtc to consider using for the connection being established.
... this example could be simplified somewhat; you may more often see the code look something like this, taking advantage of more advanced ecmascript 2016 features: let handlenewicecandidate = candidatesdp => mypeerconnection.addicecandidate(new rtcicecandidate(candidatesdp)); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecandidate.candidate' in that specification.
RTCIceCandidate.tcpType - Web APIs
"active" the transport will try to open an outbound connection but won't receive inoming connection requests.
... "passive" the transport will receive incoming connection requests but won't try to open an outbound connection.
... "so" the transport will try to open a connection simultaneously with its peer.
RTCIceCandidate - Web APIs
the rtcicecandidate interface—part of the webrtc api—represents a candidate internet connectivity establishment (ice) configuration which may be used to establish an rtcpeerconnection.
...when starting a webrtc peer connection, typically a number of candidates are proposed by each end of the connection, until they mutually agree upon one which describes the connection they decide will be best.
... webrtc then uses that candidate's details to initiate the connection.
RTCIceCandidatePair - Web APIs
the rtcicecandidatepair dictionary describes a pair of ice candidates which together comprise a description of a viable connection between two webrtc endpoints.
... properties local an rtcicecandidate describing the configuration of the local end of the connection.
... remote the rtcicecandidate describing the configuration of the remote end of the connection.
RTCIceCandidatePairStats.availableIncomingBitrate - Web APIs
the rtcicecandidatepairstats property availableincomingbitrate returns a value indicative of the available inbound capacity of the network connection represented by the candidate pair.
... syntax availableincomingbitrate = rtcicecandidatepairstats.availableincomingbitrate; value a floating-point value which approximates the amount of available bandwidth for incoming data on the network connection described by the rtcicecandidatepair.
... the value returned is calculated by adding up the available bit rate for every rtp stream using the connection described by this candidate pair.
RTCIceCandidatePairStats.availableOutgoingBitrate - Web APIs
the rtcicecandidatepairstats property availableoutgoingbitrate returns a value indicative of the available outbound capacity of the network connection represented by the candidate pair.
... syntax availableoutgoingbitrate = rtcicecandidatepairstats.availableoutgoingbitrate; value a floating-point value which approximates the amount of available bandwidth for outgoing data on the network connection described by the rtcicecandidatepair.
... the value returned is calculated by adding up the available bit rate for every rtp stream using the connection described by this candidate pair.
RTCIceCandidatePairStats.bytesReceived - Web APIs
the rtcicecandidatepairstats property bytesreceived indicates the total number of payload bytes—that is, bytes which aren't overhead such as headers or padding—that hve been received to date on the connection described by the candidate pair.
... the bytessent property reports the number of bytes sent so far on the described connection.
... syntax received = rtcicecandidatepairstats.bytesreceived; value an integer value indicating the total number of bytes received so far on the connection described by this candidate pair.
RTCIceCandidatePairStats.bytesSent - Web APIs
the rtcicecandidatepairstats property bytessent indicates the total number of payload bytes—that is, bytes which aren't overhead such as headers or padding—that hve been sent so far on the connection described by the candidate pair.
... the bytesreceived property reports the number of bytes received so far on the described connection.
... syntax sent = rtcicecandidatepairstats.bytessent; value an integer value indicating the total number of bytes sent so far on the connection described by this candidate pair.
RTCIceCandidatePairStats.currentRoundTripTime - Web APIs
the rtcicecandidatepairstats property currentroundtriptime is a floating-point value indicating the number of seconds it takes for data to be sent by this peer to the remote peer and back over the connection described by this pair of ice candidates.
... syntax rtt = rtcicecandidatepairstats.currentroundtriptime; value a floating-point value indicating the round-trip time, in seconds for the connection described by the pair of candidates for which this rtcicecandidatepairstats object offers statistics.
...this information may come from ongoing stun connectivity checks as well as from consent requests made when the connection was initially being opened.
RTCIceCandidatePairStats.nominated - Web APIs
the rtcicecandidatepairstats property nominated specifies whether or not the candidate pair described by the underlying rtcicecandidatepair has been nominated to be used as the configuration for the webrtc connection.
... syntax nominated = rtcicecandidatepairstats.nominated; value a boolean value which is set to true by the ice layer if the controlling user agent has indicated that the candidate pair should be used to configure the webrtc connection between the two peers.
... once a candidate pair has been nominated and the two peers have each reconfigured themselves to use the specified configuration, the ice negotiation process can potentially end (or it can continue, to allow the connection to adapt to changing conditions).
RTCIceCandidatePairStats.packetsReceived - Web APIs
the rtcicecandidatepairstats dictionary's packetsreceived property indicates the total number of packets of any kind that have been received on the connection described by the pair of candidates.
... the number of packets sent to date on the connection can be obtained using packetssent.
... syntax packetsreceived = rtcicecandidatepairstats.packetsreceived; value an integer value indicating the total number of packets, of any kind, which have been received on the connection described by the two candidates comprising this pair.
RTCIceCandidatePairStats.packetsSent - Web APIs
the rtcicecandidatepairstats dictionary's packetssent property indicates the total number of packets which have been sent on the connection described by the pair of candidates.
... the number of packets received to date on the connection can be obtained using packetsreceived.
... syntax packetssent = rtcicecandidatepairstats.packetssent; value an integer value indicating the total number of packets, of any kind, which have been sent on the connection described by the two candidates comprising this pair.
RTCIceCandidatePairStats.readable - Web APIs
the obsolete rtcicecandidatepairstats property readable reports whether or not the connection described by the candidate pair has received at least one valid incoming ice request.
... syntax isreadable = rtcicecandidatepairstats.readable; value a boolean value which is true if the connection described by this candidate pair has received at least one valid ice request, and is therefore ready to be read from.
... note: this property was removed from the specification in early 2017 because you can determine whether or not the connection is readable by checking to see if requestsreceived is greater than 0: if (icpstats.requestsreceived > 0) { /* at least one ice request has been received */ } ...
RTCIceServer.url - Web APIs
WebAPIRTCIceServerurl
the obsolete rtciceserver dictionary's url property specifies the url of a single ice server to be used while negotiating connections.
... example this example creates a new rtcpeerconnection which will use a stun server at stunserver.example.org to negotiate connections.
... mypeerconnection = new rtcpeerconnection({ iceservers: [ { url: "stun:stunserver.example.org" } ] }); unfortunately, the only way to tell ice that the server has a backup domain name of stunserver2.example.org is to add a new entry to the iceservers array for it.
RTCIceServer - Web APIs
objects of this type are provided in the configuration of an rtcpeerconnection, in the iceservers array.
... avoid specifying an unnecessarily large number of urls in the urls property; the startup time for your connection will go up substantially.
... "stun:stun.services.mozilla.com", username: "louis@mozilla.com", credential: "webrtcdemo" }, { urls: [ "stun:stun.example.com", "stun:stun-1.example.com" ] }] }; var pc = new rtcpeerconnection(configuration); once the configuration object has been created, it is passed into the rtcpeerconnection() constructor to use it as the configuration for the new peer connection.
RTCIceTcpCandidateType - Web APIs
values "active" the transport will try to open an outbound connection but won't receive inoming connection requests.
... "passive" the transport will receive incoming connection requests but won't try to open an outbound connection.
... "so" the transport will try to open a connection simultaneously with its peer.
RTCIceTransport.getSelectedCandidatePair() - Web APIs
local describes the configuration of the local end of the connection, while remote describes the remote peer's configuration.
... usage notes as the ice agent performs negotiation of a rtcpeerconnection, it gathers and analyzes candidate configurations from each the two peers.
... as soon as it finds an acceptable matching pair of candidates, meeting the requirements for the connection, a selectedcandidatepairchange event is fired at the rtcicetransport.
RTCIceTransport.state - Web APIs
"checking" at least one remote candidate has been received, and the rtcicetransport has begun examining pairings of remote and local candidates in order to attempt to identify viable pairs that could be used to establish a connection.
...a value of "disconnected" means that a transient issue has occurred that has broken the connection, but that should resolve itself automatically without your code having to take any action.
...this is a terminal state, indicating that the connection cannot be achieved or maintained.
RTCIdentityErrorEvent - Web APIs
this is usually for an rtcpeerconnection.
... firefox implements this interface under the following name: rtcpeerconnectionidentityerrorevent.
... it is likely that it will correct this name when it will unprefix rtcpeerconnection, once spec and implementation will have been stabilized.
RTCIdentityEvent - Web APIs
this is usually for an rtcpeerconnection.
... firefox implements this interface under the following name: rtcpeerconnectionidentityevent.
... it is likely that it will correct this name when it will unprefix rtcpeerconnection, once spec and implementation will have been stabilized.
RTCRtpSender.dtmf - Web APIs
WebAPIRTCRtpSenderdtmf
the read-only dtmf property on the rtcrtpsender interface returns a rtcdtmfsender object which can be used to send dtmf tones over the rtcpeerconnection .
... syntax var dtmfsender = rtcrtpsender.dtmf; value an rtcdtmfsender which can be used to send dtmf over the rtp session, or null if the track being carried by the rtp session or the rtcpeerconnection as a whole doesn't support dtmf.
... only audio tracks can support dtmf, and typically only one audio track per rtcpeerconnection will have an associated rtcdtmfsender example tbd specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpsender.dtmf' in that specification.
RTCRtpTransceiver.setCodecPreferences() - Web APIs
the specified set of codecs and configurations will be used for all future connections including this transceiver until this method is called again.
... when preparing to open an rtcpeerconnection, you can change the codec parameters from the user agent's default configuration by calling setcodecparameters() before calling either rtcpeerconnection.createoffer() or createanswer().
... note: any codecs not included in codecs will not be considered during the process of negotiating a connection.
RTCSctpTransport.state - Web APIs
its value is one of the following: connecting the initial state when the connection is being estabilished.
... connected the connection is open for data transmission.
... closed the connection is closed and can no longer be used.
RTCSctpTransport - Web APIs
this provides information about limitations of the transport, but also provides a way to access the underlying datagram transport layer security (dtls) transport over which sctp packets for all of an rtcpeerconnection's data channels are sent and received.
... you don't create rtcsctptransport objects yourself; instead, you get access to the rtcsctptransport for a given rtcpeerconnection through its sctp property.
... possibly the most useful property on this interface is its maxmessagesize property, which you can use to determine the upper limit on the size of messages you can send over a data channel on the peer connection.
RTCSessionDescription - Web APIs
the rtcsessiondescription interface describes one end of a connection—or potential connection—and how it's configured.
... the process of negotiating a connection between two peers involves exchanging rtcsessiondescription objects back and forth, with each description suggesting one combination of connection configuration options that the sender of the description supports.
... once the two peers agree upon a configuration for the connection, negotiation is complete.
RTCTrackEvent() - Web APIs
the rtctrackevent() constructor creates and returns a new rtctrackevent object, configured to describe the track which has been added to the rtcpeerconnection.
... syntax trackevent = new rtctrackevent(eventinfo); parameters eventinfo an object based on the rtctrackeventinit dictionary, providing information about the track which has been added to the rtcpeerconnection.
... return value a new rtctrackevent describing a track which has been added to the rtcpeerconnection.
WebSocket: close event - Web APIs
the close event is fired when a connection with a websocket is closed.
... bubbles no cancelable no interface closeevent event handler property onclose examples you might want to know when the connection has been closed so that you can update the ui or, perhaps, save data about the closed connection.
... examplesocket.addeventlistener('close', (event) => { console.log('the connection has been closed successfully.'); )}; you can perform the same actions using the event handler property, like this: examplesocket.onclose = function (event) { console.log('the connection has been closed successfully.'); }; specifications specification status html living standardthe definition of 'websocket close' in that specification.
WebSocket: open event - Web APIs
the open event is fired when a connection with a websocket is opened.
... bubbles no cancelable no interface event event handler property onopen examples // create websocket connection.
... const socket = new websocket('ws://localhost:8080'); // connection opened socket.addeventlistener('open', (event) => { socket.send('hello server!'); }); specifications specification status html living standardthe definition of 'websocket open' in that specification.
WebSocket.send() - Web APIs
WebAPIWebSocketsend
the websocket.send() method enqueues the specified data to be transmitted to the server over the websocket connection, increasing the value of bufferedamount by the number of bytes needed to contain the data.
... exceptions thrown invalid_state_err the connection is not currently open.
... note: gecko's implementation of the send() method differs somewhat from the specification in gecko 6.0; gecko returns a boolean indicating whether or not the connection is still open (and, by extension, that the data was successfully queued or transmitted); this is corrected in gecko 8.0.
Window - Web APIs
WebAPIWindow
globaleventhandlers.onmouseover called when the pointer enters the window globaleventhandlers.onmouseup called when any mouse button is released windoweventhandlers.onoffline called when network connection is lost.
... windoweventhandlers.ononline called when network connection is established.
... connection events offline fired when the browser has lost access to the network and the value of navigator.online has switched to false.
Using the application cache - HTML: Hypertext Markup Language
network: files listed under the network: section header in the cache manifest file are white-listed resources that require a connection to the server.
...fallback.html this example uses network and fallback sections to specify that the network.html page must always be retrieved from the network, and that the fallback.html page should be served as a fallback resource (e.g., in case a connection to the server cannot be established).
...in the fallback section each line is a valid uri or iri reference to a resource, followed by a fallback resource that is to be served up when a connection with the server cannot be made.
CONNECT - HTTP
WebHTTPMethodsCONNECT
the client asks an http proxy server to tunnel the tcp connection to the desired destination.
... the server then proceeds to make the connection on behalf of the client.
... once the connection has been established by the server, the proxy server continues to proxy the tcp stream to and from the client.
Web video codec guide - Web media technologies
however, mp4v-es is also used to transmit mpeg-4 audio and video over a mobile connection using 3gp.
... [2] safari only supports vp8 in webrtc connections.
...that's still a huge number of bits to pump through a connection every second, and is not currently practical for any real-world use.
Security best practices in extensions - Archive of obsolete content
sandboxed http connections the main purpose of sandboxed http connections is to interact with a web service, without interfering with the cookies set in the browser by that service/site.
... for example, if you are loading pictures or other data from a photo sharing site, you can sandbox your connections to that site so that the normal browsing of that site by the user in the main firefox browser is not affected.
Setting up an extension development environment - Archive of obsolete content
devtools.debugger.prompt-connection = false.
... this prevents the browser debugger from prompting for connection permission every time.
Promises - Archive of obsolete content
representative example usage components.utils.import("resource://gre/modules/sqlite.jsm"); task.spawn(function* () { // open the connection.
... let db = yield sqlite.openconnection({ path: database_path }); try { // start a transaction to insert the data.
Monitoring downloads - Archive of obsolete content
note: the mozistorageconnection method close() is being added to firefox 3 alpha 8; in prior versions of firefox, there is no way to explicitly close the database.
... instead, it is closed when the garbage collector disposes of the connection object.
Mozilla Application Framework in Detail - Archive of obsolete content
more information using the mozilla soap api (an article written for the apple developer connection) xml extras (the home page for the xml extras code module, which contains mozilla's web services support) soap scripts in mozilla (documentation on soap in mozilla from the engineer who implemented it) xpinstall, mozilla's cross platform installation technology one of the many things that makes the mozilla platform easy for both users and developers is that applications can be installed, exten...
...a runtime engine that provides platform-independence (across over a dozen platforms) for non-gui operating system facilities with support for threads, thread synchronization, normal file and network i/o, interval timing and calendar time, basic memory management (malloc and free) and shared library linking; psm, a set of libraries that perform cryptographic operations including setting up an ssl connection, object signing and signature verification, certificate management (including issuance and revocation), other common pki functions, and s/mime support; an sql support that provides the ability to set up data sources, query a database, and retrieve results as javascript objects or rdf data sources; and an api for directory services via the lightweight directory access protocol (ldap).
Proxy UI - Archive of obsolete content
products firefox menu: preferences > advanced tab > networking tab > connection button.
... thunderbird menu: preferences > advanced tab > networking tab > connection button.
The Implementation of the Application Object Model - Archive of obsolete content
what we really need in this situation is the ability to take our intermediate representation and form an entirely different set of connections between our data objects.
...this implies a need for the ability to make "negative" and "positive" assertions about connections in our tree, i.e., so that we can delete arcs and/or add arcs to the tree.
Introduction to Public-Key Cryptography - Archive of obsolete content
the client sends the name and password across the network, either in the clear or over an encrypted ssl connection.
...the source can be either the verifier's local certificate database (on that client or server) or the certificate chain provided by the subject (for example, over an ssl connection).
WebRTC data channels - Game development
a webrtc data channel lets you send text or binary data over an active connection to a peer.
...this library provides a simple api for creating peer connections and setting up streams and data channels.
Visual typescript game engine - Game development
networking is based on websocket full-duplex communication only.you must be conform with classic socket connection methodology.
...ense ├── build/ (this is auto generated) | ├── externals/ | ├── templates/ | ├── imgs/ | ├── styles/ | | └── favicon.ico | ├── visualjs2.js | ├── app.html ├── src/ | ├── style/ | | ├── styles.css | ├── libs/ | | ├── class/ | | | ├── networking/ | | | | ├── rtc-multi-connection/ | | | | | ├── filebufferreader.js | | | | | ├── rtcmulticonnection2.js | | | | | ├── rtcmulticonnection3.js | | | | | ├── linkify.js | | | | | ├── gethtmlmediaelement.js | | | | | ├── socket.io.js | | | | ├── broadcaster-media.ts | | | | ├── broadcaster.ts | | | | �...
ICE - MDN Web Docs Glossary: Definitions of Web-related terms
this protocol lets two peers find and establish a connection with one another even though they may both be using network address translator (nat) to share a global ip address with other devices on their respective local networks.
... the framework algorithm looks for the lowest-latency path for connecting the two peers, trying these options in order: direct udp connection (in this case—and only this case—a stun server is used to find the network-facing address of a peer) direct tcp connection, via the http port direct tcp connection, via the https port indirect connection via a relay/turn server (if a direct connection fails, e.g., if one peer is behind a firewall that blocks nat traversal) learn more general knowledge webrtc, the principal web-related protocol which uses ice webrtc protocols technical reference rfc 5245, the ietf specification for ice rtcicecandidate, the interface representing a ice candidate ...
Session Hijacking - MDN Web Docs Glossary: Definitions of Web-related terms
break the victim machine's connection.
... protection against session hijacking create a secure communication channel with ssh (secure shell) pass authentication cookies over https connection implement logout functionality so the user can end the session generate the session id after successful login pass encrypted data between the users and the web server use a string or long random number as a session key learn more general knowledge session hijacking on wikipedia ...
Transport Layer Security (TLS) - MDN Web Docs Glossary: Definitions of Web-related terms
all modern browsers support the tls protocol, requiring the server to provide a valid digital certificate confirming its identity in order to establish a secure connection.
...from version 74 onwards, firefox will return a secure connection failed error when connecting to servers using the older tls versions (bug 1606734).
WebRTC - MDN Web Docs Glossary: Definitions of Web-related terms
webrtc consists mainly of these parts: getusermedia() grants access to a device's camera and/or microphone, and can plug in their signals to a rtc connection.
... rtcpeerconnection an interface to configure video chat or voice calls.
What is accessibility? - Learn web development
we traditionally think of this as being about people with disabilities, but the practice of making sites accessible also benefits other groups such as those using mobile devices, or those with slow network connections.
...this benefits everyone, especially those on mobile devices and/or slow connections.
Basic native form controls - Learn web development
the best way to protect users from this is to host any pages involving forms over a secure connection (i.e.
... browsers recognize the security implications of sending form data over an insecure connection, and have warnings to deter users from using insecure forms.
How the Web works - Learn web development
in addition to the client and the server, we also need to say hello to: your internet connection: allows you to send and receive data on the web.
...this message, and all other data sent between the client and the server, is sent across your internet connection using tcp/ip.
Graceful asynchronous programming with Promises - Learn web development
the code that the video chat application would use might look something like this: function handlecallbutton(evt) { setstatusmessage("calling..."); navigator.mediadevices.getusermedia({video: true, audio: true}) .then(chatstream => { selfviewelem.srcobject = chatstream; chatstream.gettracks().foreach(track => mypeerconnection.addtrack(track, chatstream)); setstatusmessage("connected"); }).catch(err => { setstatusmessage("failed to connect"); }); } this function starts by using a function called setstatusmessage() to update a status display with the message "calling...", indicating that a call is being attempted.
... it then calls getusermedia(), asking for a stream that has both video and audio tracks, then once that's been obtained, sets up a video element to show the stream coming from the camera as a "self view," then takes each of the stream's tracks and adds them to the webrtc rtcpeerconnection representing a connection to another user.
What is web performance? - Learn web development
when you request a url and hit enter / return, the browser finds out where the server is that holds that website's files, establishes a connection to it, and requests the files.
...there is overhead involved in establishing tcp and http connections, and some unavoidable latency in pushing the request and response bytes back and forth across the network, but there are certain ways to reduce latency (e.g.
Old Thunderbird build
good internet connection for the initial source download.
...first, cd into the comm-central subdirectory (created automatically by the previous command): cd comm-central then run: python client.py checkout on some types of network connections, "hg clone" might fail because it gets interrupted.
Simple Instantbird build
good internet connection for the initial source download.
...first, cd into the comm-central subdirectory (created automatically by the previous command): cd comm-central then run: python client.py checkout note: unless you have a very good network connection, "hg clone" might fail because it gets interrupted.
Getting Started with Chat
you will need to use the following information to configure the server connection: server: irc.mozilla.org port: 6667 (default) or 6697 (ssl) desktop clients desktop clients tens to allow the most detailed configuration.
... for example, you can connect using ssl or a non-secure connection, and you can configure the client to automatically connect to a server and join a particular set of channels when you start the client.
IPDL Tutorial
amplechild* toplevelchild::allocpexample() { refptr<examplechild*> actor = new examplechild(); return actor.forget(); } virtual bool toplevelchild::deallocpexample(pexamplechild* actor) { ns_release(static_cast<examplechild*>(actor)); return true; } if an object that implements a protocol can't be constructed inside allocpfoo, has been previously constructed and doesn't require an ipdl connection throughout its lifetime, or implements a refcounted protocol where the first form of constructor is not available, there is a second form of sendpfooconstructor which can be used: class examplechild { public: void dosomething() { amanagerchild->sendpexampleconstructor(this, ...); } }; internally, the first constructor form simply calls pexample(parent|child)* actor = allocpexa...
... when a manager protocol is destroyed, any subprotocols will be notified: no further messages will be accepted deallocpsubprotocol will be called on the manager protocol to deallocate any active subprotocols when the toplevel protocol is destroyed, this is equivalent to shutting down the entire ipdl machinery for that connection, because no more messages can be sent and all subprotocols are destroyed.
Bootstrapping a new locale
--> after the localization notes, you will see a list of <!entity> strings like the following: <!entity certerror.pagetitle "untrusted connection"> you should go through each entity, translating the value in the parameters (e.g.
... untrusted connection in the example above), like so (example for polish): <!entity certerror.pagetitle "niezaufane połączenie"> once you have translated all the <!entity> strings in this file, you should save your work and open the next .dtd or .property file.
Localizing without a specialized tool
--> after the localization notes, you will see a list of <!entity> strings like the following: <!entity certerror.pagetitle "untrusted connection"> you should go through each entity, translating the value in the parameters (e.g.
... untrusted connection in the example above), like so (example for polish): <!entity certerror.pagetitle "niezaufane połączenie"> once you have translated all the <!entity> strings in this file, you should save your work and open the next .dtd or .properties file.
Nonblocking IO In NSPR
the tcp connection on that socket has been closed (end of stream).
...if <tt>pr_poll()</tt> reports that the socket is readable (i.e., <tt>pr_poll_read</tt> is set in <tt>out_flags</tt>), and <tt>pr_available()</tt> returns 0, this means that the socket connection is closed.
NSS 3.28.1 release notes
9d:57:1b:c5:92 cn = symantec class 2 public primary certification authority - g6 sha-256 fingerprint: cb:62:7d:18:b5:8a:d5:6d:de:33:1a:30:45:6b:c6:5c:60:1a:4e:9b:18:de:dc:ea:08:e7:da:aa:07:81:5f:f0 the version number of the updated root ca list has been set to 2.11 a misleading assertion/alert has been removed, when nss tries to flush data to the peer but the connection was already reset.
... bugs fixed in nss 3.28.1 bug 1296697 - december 2016 batch of root ca changes bug 1322496 - internal error assert when the other side closes connection before reading eoed compatibility nss 3.28.1 shared libraries are backward compatible with all older nss 3.x shared libraries.
NSS 3.35 release notes
tls servers can screen new tls 1.3 connections, as they are made using the experimental ssl_helloretryrequestcallback function.
... the application is then able to examine application-chosen content from the session tickets, or helloretryrequest cookie, and decide whether to proceed with the connection.
NSS Tools sslstrength
the enabled ciphersuites will always be printed out before the connection is made.
... step-up step up is a mode where the connection starts out with 40-bit encryption, but due to a 'change-cipher-spec' handshake, changes to 128-bit encryption.
NSS environment variables
in other words a connection will be dropped at initial handshake if a server or client do not support safe renegotiation.
...this allows packet sniffers to decrypt tls connections.
NSS Tools sslstrength
the enabled ciphersuites will always be printed out before the connection is made.
... step-up step up is a mode where the connection starts out with 40-bit encryption, but due to a 'change-cipher-spec' handshake, changes to 128-bit encryption.
Necko Interfaces Overview
scheme, host, path, ...) per protocol implementation necko provides uri impl's for common uri formats (see nsstandardurl, nssimpleuri) nsichannel : nsirequest represents a logical connection to the resource identified by a nsiuri per protocol implementation single use (ie.
...quest channel impl adds itself to its load group during invocation of asyncopen channel impl removes itself from its load group when download completes load groups in gecko own all channels used to load a particular page (until the channels complete) all channels owned by a load group can be canceled at once via the load group's nsirequest::cancel method nsitransport represents a physical connection, such as a file descriptor or a socket used directly by protocol handler implementations (as well as by mailnews and chatzilla) synchronous i/o methods: openinputstream, openoutputstream asynchronous i/o methods: asyncread, asyncwrite nsitransport::asyncread takes a nsistreamlistener parameter original document information author(s): darin fisher last updated date: december 10, 2001 ...
Redis Tips
so rather than thinking about redis as a database with some kind of non-existent relationship to sql, think of it as a data structure server with a rich set of commands for querying and manipulating those data structures over a network connection.
... here's a stupid node script to show how this works: #!/usr/bin/env node var r = require('redis').createclient(); r.multi() .set("foo", 42) .set("bar", "ice cream") .set("baz", 6.28) .get("foo") .get("bar") .get("baz") .exec(function(err, resultlist) { console.log(json.stringify(resultlist, null, 2)); r.end(); // terminate the redis connection; node can quit }); when run, this prints: [ "ok", "ok", "ok", "42", "ice cream", "6.28" ] the result list includes one value per each command executed.
Places Developer Guide
here is how one can get a connection to the places database: function getplacesdbconn() { return components.classes['@mozilla.org/browser/nav-history-service;1'].
... getservice(components.interfaces.nspiplacesdatabase).dbconnection; } and then to get the a redirected visit_id from another visit_id: function getfromvisit(visit_id) { var sql = <cdata><![cdata[ select from_visit from moz_places, moz_historyvisits where moz_historyvisits.id = :visit_id and moz_places.id = moz_historyvisits.place_id; ]]></cdata>.tostring(); var sql_stmt = getplacesdbconn.createstatement(sql); sql_stmt.params.visit_id = visit_id; var from_visit; try { // here we can't use the "executeasync" method since have to return a // result right-away.
mozIStorageVacuumParticipant
method overview boolean onbeginvacuum(); void onendvacuum(in boolean asucceeded); attributes attribute type description databaseconnection mozistorageconnection a connection to the database file to be vacuumed.
...the recommended value is mozistorageconnection::default_page_size.
nsIAuthPrompt2
level_pw_encrypted 1 password will be sent encrypted, but the connection is otherwise insecure.
... level_secure 2 the connection, both for password and data, is secure.
nsICookieManager2
aissecure true if the cookie should only be sent over a secure connection.
... aishttponly true if the cookie should only be sent to, and can only be modified by, an http connection.
nsIDNSService
to access the service, use: var dnsservice = components.classes["@mozilla.org/network/dns-service;1"] .createinstance(components.interfaces.nsidnsservice); note: starting in gecko 7.0, the "happy eyeballs" strategy is used to reduce lengthy timeouts when attempting backup connections during attempts to connect from clients that have broken ipv6 connectivity.
... this is done by disabling ipv6 on backup connections.
nsIDOMMozNetworkStatsData
1.0 66 introduced gecko 18.0 inherits from: nsisupports last changed in gecko 18.0 (firefox 18.0 / thunderbird 18.0 / seamonkey 2.15) attributes attribute type description rxbytes unsigned long the number of bytes received on the connection.
... txbytes unsigned long the number of bytes transmitted on the connection.
nsIHttpUpgradeListener
void ontransportavailable( in nsisockettransport atransport, in nsiasyncinputstream asocketin, in nsiasyncoutputstream asocketout ); parameters atransport the nsisockettransport describing the socket connection between the browser and the server; this socket can now be used for the new protocol instead of http.
... asocketin the nsiasyncinputstream object representing the input stream for data coming from the server over the socket connection.
nsIMsgIncomingServer
inherits from: nsisupports last changed in gecko 1.9.1 (firefox 3.5 / thunderbird 3.0 / seamonkey 2.0) method overview void clearallvalues(); void cleartemporaryreturnreceiptsfilter(); void closecachedconnections(); void configuretemporaryfilters(in nsimsgfilterlist filterlist); void configuretemporaryreturnreceiptsfilter(in nsimsgfilterlist filterlist); obsolete since gecko 1.8 void displayofflinemsg(in nsimsgwindow awindow); boolean equals(in nsimsgincomingserver server); void forgetpassword(); void forgetsessionpassword(); astring generateprettynameformigration(); boolean getboolattribute(in string name); boolean getboolvalue(in string at...
...exceptions thrown missing exception missing description closecachedconnections() void closecachedconnections(); parameters none.
nsIServerSocketListener
netwerk/base/public/nsiserversocket.idlscriptable this interface is notified whenever a server socket accepts a new connection.
... inherits from: nsisupports last changed in gecko 1.7 method overview void onsocketaccepted(in nsiserversocket aserv, in nsisockettransport atransport); void onstoplistening(in nsiserversocket aserv, in nsresult astatus); methods onsocketaccepted() this method is called when a client connection is accepted.
nsISpeculativeConnect
netwerk/base/public/nsispeculativeconnect.idlscriptable lets non-networking code provide hints to the networking layer that an http connection attempt to a particular site is likely to happen soon; this lets the networking layer begin setting up tcp and, if appropriate, ssl handshakes to save time when the connection is actually opened later.
... 1.0 66 introduced gecko 15.0 inherits from: nsisupports last changed in gecko 15.0 (firefox 15.0 / thunderbird 15.0 / seamonkey 2.12) to use this service, simply call nsisupports.queryinterface() on the nsiioservice you plan to use for the connection, to get access to the i/o service's implementation of nsispeculativeconnect.
Use SQLite
const cc = components.classes; const ci = components.interfaces; var tbirdsqlite = { onload: function() { // initialization code this.initialized = true; this.dbinit(); }, dbconnection: null, dbschema: { tables: { attachments:"id integer primary key, \ name text \ encoded text not null" } }, dbinit: function() { var dirservice = cc["@mozilla.org/file/directory_service;1"].
... getservice(ci.mozistorageservice); var dbconnection; if (!dbfile.exists()) dbconnection = this._dbcreate(dbservice, dbfile); else { dbconnection = dbservice.opendatabase(dbfile); } this.dbconnection = dbconnection; }, _dbcreate: function(adbservice, adbfile) { var dbconnection = adbservice.opendatabase(adbfile); this._dbcreatetables(dbconnection); return dbconnection; }, _dbcreatetables: function(adbconnection) { for(var name in this.dbschema.tables) adbconnection.createtable(name, this.dbschema.tables[name]); }, }; window.addeventlistener("loa...
Index - Firefox Developer Tools
47 inspecting web sockets javascript, network monitor, webapis, web socket inspector, web sockets since firefox 71, the network monitor has had the ability to inspect web socket connections.
... 53 throttling 110n:priority, debugging, dev tools, firefox, guide, networking, tools the network monitor allows you to throttle your network speed to emulate various connection speeds so you can see how your app will behave under different connection types.
Network request list - Firefox Developer Tools
if the request used ssl/tls and the connection had security weaknesses such as weak ciphers, you'll see a warning triangle next to the domain.
... websocket connections use the ws button in the toolbar.
Responsive Design Mode - Firefox Developer Tools
dpr (pixel ratio) - beginning with firefox 68, the dpr is no longer editable; create a custom device in order to change the dpr throttling - a drop-down list where you can select the connection throttling to apply, for example 2g, 3g, or lte enable/disable touch simulation - toggles whether or not responsive design mode simulates touch events.
... network throttling if you do all your development and testing using a very fast network connection, users may experience problems with your site if they are using a slower connection.
AudioNode.channelCountMode - Web APIs
the possible values of channelcountmode and their meanings are: value description the following audionode children default to this value max the number of channels is equal to the maximum number of channels of all connections.
... gainnode, delaynode, scriptprocessornode, channelmergernode, biquadfilternode, waveshapernode clamped-max the number of channels is equal to the maximum number of channels of all connections, clamped to the value of channelcount.
AudioNode.connect() - Web APIs
WebAPIAudioNodeconnect
notsupportederror the specified connection would create a cycle (in which the audio loops back through the same nodes repeatedly) and there are no delaynodes in the cycle to prevent the resulting waveform from getting stuck constructing the same audio frame indefinitely.
...next, it will mix it together with any other such outputs, and the intrinsic parameter value (the value the audioparam would normally have without any audio connections), including any timeline changes scheduled for the parameter.
Applying styles and colors - Web APIs
for lines which are at large angles with each other, this point is not far from the inside connection point.
... the miterlimit property determines how far the outside connection point can be placed from the inside connection point.
EventSource.onopen - Web APIs
the onopen property of the eventsource interface is an eventhandler called when an open event is received, that is when the connection was just opened.
... syntax eventsource.onopen = function examples evtsource.onopen = function() { console.log("connection to server opened."); }; note: you can find a full example on github — see simple sse demo using php.
EventSource: open event - Web APIs
the open event of the eventsource api is fired when a connection with an event source is opened.
... bubbles no cancelable no interface event event handler property eventsource.onopen examples var evtsource = new eventsource('sse.php'); // addeventlistener version evtsource.addeventlistener('open', (e) => { console.log("the connection has been established."); }); // onopen version evtsource.onopen = (e) => { console.log("the connection has been established."); }; specifications specification status html living standardthe definition of 'open event' in that specification.
EventSource.readyState - Web APIs
the readystate read-only property of the eventsource interface returns a number representing the state of the connection.
... syntax var myreadystate = eventsource.readystate; value a number representing the state of the connection.
Using the Gamepad API - Web APIs
using button information let's look at a simple example that displays connection information for one gamepad (it ignores subsequent gamepad connections) and allows you to move a ball around the screen using the four gamepad buttons on the right hand side of the gamepad.
... to start with, we declare some variables: the gamepadinfo paragraph that the connection info is written into, the ball that we want to move, the start variable that acts as the id for requestanimation frame, the a and b variables that act as position modifiers for moving the ball, and the shorthand variables that will be used for the requestanimationframe() and cancelanimationframe() cross browser forks.
IDBFactorySync - Web APIs
method overview idbdatabasesync open (in domstring name, in domstring description, in optional boolean modifydatabase) raises (idbdatabaseexception); methods open() opens and returns a connection to a database.
... blocks the calling thread until the connection object is ready to return.
Using IndexedDB - Web APIs
once all of the transactions have completed, the database connection is closed.
... finally, the idbdatabase object representing the database connection receives a close event.
IndexedDB API - Web APIs
you need to specify the database schema, open a connection to your database, and then retrieve and update data within a series of transactions.
... idbdatabase represents a connection to a database.
MediaTrackSettings.echoCancellation - Web APIs
echo cancellation is a feature which attempts to prevent echo effects on a two-way audio connection by attempting to reduce or eliminate crosstalk between the user's output device and their input device.
... because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
Media Capture and Streams API (Media Stream) - Web APIs
a non-local mediastream may be representing to a media element, like <video> or <audio>, a stream originating over the network, and obtained via the webrtc rtcpeerconnection api, or a stream created using the web audio api mediastreamaudiosourcenode.
...it can be a media elements, like <audio> or <video>, the webrtc rtcpeerconnection api or a web audio api mediastreamaudiosourcenode.
MutationObserver.observe() - Web APIs
observation follows nodes when disconnected mutation observers are intended to let you be able to watch the desired set of nodes over time, even if the direct connections between those nodes are severed.
...this prevents you from missing changes that occur after the connection is severed and before you have a chance to specifically begin monitoring the moved node or subtree for changes.
Navigator.onLine - Web APIs
for example, the property should return false when users click links soon after they lose internet connection.
...until firefox 41, all other conditions return a true value; testing actual behavior on nightly 68 on windows shows that it only looks for lan connection like chrome and safari giving false positives.
NetworkInformation.downlink - Web APIs
this value is based on recently observed application layer throughput across recently active connections, excluding connections made to a private address space.
... in the absence of recent bandwidth measurement data, the attribute value is determined by the properties of the underlying connection technology.
PerformanceResourceTiming.requestStart - Web APIs
if the transport connection fails and the browser retires the request, the value returned will be the start of the retry request.
...print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
RTCDTMFSender.insertDTMF() - Web APIs
the insertdtmf() method on the rtcdtmfsender interface starts sending dtmf tones to the remote peer over the rtcpeerconnection.
... as long as the connection is active, you can send tones at any time.
RTCDTMFSender - Web APIs
the rtcdtmfsender interface provides a mechanism for transmitting dtmf codes on a webrtc rtcpeerconnection.
... you gain access to the connection's rtcdtmfsender through the rtcrtpsender.dtmf property on the audio track you wish to send dtmf with.
RTCDataChannel.close() - Web APIs
most of the process of closing the connection is handled asynchronously; you can detect when the channel has finished closing by watching for a close event on the data channel.
... example var pc = new rtcpeerconnection(); var dc = pc.createdatachannel("my channel"); dc.onmessage = function (event) { console.log("received: " + event.data); dc.close(); // we decided to close after the first received message }; dc.onopen = function () { console.log("datachannel open"); }; dc.onclose = function ( console.log("datachannel close"); }; // now negotiate the connection and so forth...
RTCDataChannel.id - Web APIs
WebAPIRTCDataChannelid
each rtcpeerconnection can therefore have up to a theoretical maximum of 65,534 data channels on it, although the actual maximum may vary from browser to browser.
... example var pc = new rtcpeerconnection(); var dc = pc.createdatachannel("my channel"); console.log("channel id: " + dc.id); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.id' in that specification.
RTCDataChannel.maxRetransmits - Web APIs
this can only be set when the rtcdatachannel is created by calling rtcpeerconnection.createdatachannel(), using the maxretransmits field in the specified options.
... syntax var tries = adatachannel.maxretransmits; value the maximum number of times the browser will try to retransmit a message before giving up, or null if not set when rtcpeerconnection.createdatachannel() was called.
RTCDataChannel.onclose - Web APIs
this way, the message entry field and the send button are only enabled for use when the connection is actually open.
... let pc = new rtcpeerconnection(); let dc = pc.createdatachannel("messagechannel")}}; dc.onopen = function(event) { document.getelementbyid("messagebox").disabled = false; document.getelementbyid("sendbutton").disabled = false; }; dc.onclose = function(event) { document.getelementbyid("messagebox").disabled = true; document.getelementbyid("sendbutton").disabled = true; } /* now negotiate the connection, etc...
RTCDataChannel.onmessage - Web APIs
example this code snippet creates a peer connection, adds a data channel to it, and starts creating new <p> (paragraph) elements each time a message arrives, with the message's contents displayed inside it.
... let pc = new rtcpeerconnection(); let dc = pc.createdatachannel(); dc.onmessage = function(event) { var el = document.createelement("p"); var txtnode = document.createtextnode(event.data); el.appendchild(txtnode); receivebox.appendchild(el); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.onmessage' in that specification.
RTCDataChannel.onopen - Web APIs
example this example adds a new data channel to an existing rtcpeerconnection, mypeerconnection.
... let dc = mypeerconnection.createdatachannel("message channel"); dc.onopen = function(event) { let messagebox = document.getelementbyid("messagebox"); let sendbutton = document.getelementbyid("sendbutton"); messagebox.disabled = false; messagebox.focus(); sendbutton.disabled = false; } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.onopen' in that specification.
RTCDataChannel.ordered - Web APIs
this is set when the rtcdatachannel is created, by setting the ordered property on the rtcdatachannelinit object passed as rtcpeerconnection.createdatachannel()'s options parameter.
... example var pc = new rtcpeerconnection(); var dc = pc.createdatachannel("my channel"); if (!dc.ordered) { // handle unordered messaging } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.ordered' in that specification.
RTCError - Web APIs
WebAPIRTCError
datachannel.addeventlistener("error", (event) => { let error = event.error; if (error.errordetail === "sdp-syntax-error") { let errline = error.sdplinenumber; let errmessage = error.message; let alertmessage = `a syntax error occurred interpreting line ${errline} of the sdp: ${errmessage}`; showmyalertmessage("data channel error", alertmessage); } else { terminatemyconnection(); } }); if the error is an sdp syntax error—indicated by its errordetail property being sdp-syntax-error—, a message string is constructed to present the error message and the line number within the sdp at which the error occurred.
... any other error is treated as terminal, causing a terminatemyconnection() function to be called.
RTCErrorEvent.error - Web APIs
datachannel.addeventlistener("error", (event) => { let error = event.error; if (error.errordetail === "sdp-syntax-error") { let errline = error.sdplinenumber; let errmessage = error.message; let alertmessage = `a syntax error occurred interpreting line ${errline} of the sdp: ${errmessage}`; showmyalertmessage("data channel error", alertmessage); } else { terminatemyconnection(); } }); if the error is an sdp syntax error—indicated by its errordetail property being sdp-syntax-error—, a message string is constructed to present the error message and the line number within the sdp at which the error occurred.
... any other error is treated as terminal, causing a terminatemyconnection() function to be called.
RTCIceCandidatePair.local - Web APIs
the local property of the rtcicecandidatepair dictionary specifies the rtcicecandidate which describes the configuration of the local end of a viable webrtc connection.
... var candidatepair = pc.getsenders()[0].transport.transport.getselectedcandidatepair(); var localcandidate = candidatepair.local; the rtcicetransport is found by getting the list of rtcrtpsender objects for the rtcpeerconnection pc.
RTCIceCandidatePair.remote - Web APIs
the remote property of the rtcicecandidatepair dictionary specifies the rtcicecandidate describing the configuration of the remote end of a viable webrtc connection.
... var candidatepair = pc.getsenders()[0].transport.transport.getselectedcandidatepair(); var remotecandidate = candidatepair.remote; the rtcicetransport is found by getting the list of rtcrtpsender objects for the rtcpeerconnection pc.
RTCIceCandidatePairStats.consentRequestsSent - Web APIs
the rtcicecandidatepairstats property consentrequestssent specifies the number of consent requests that have been sent by this peer to the remote peer on the connection described by the pair of candidates.
... syntax consentrequestssent = rtcicecandidatepairstats.consentrequestssent; value an integer indicating the number of consent requests this peer has sent to the other peer over the connection described by the pair of candidates referenced by this rtcicecandidatepairstats object.
RTCIceCandidatePairStats.lastPacketReceivedTimestamp - Web APIs
the rtcicecandidatepairstats property lastpacketreceivedtimestamp indicates the time at which the connection described by the candidate pair last received a packet.
... syntax lastpacketreceivedtimestamp = rtcicecandidatepairstats.lastpacketreceivedtimestamp; value a domhighrestimestamp object indicating the timestamp at which the connection described by pair of candidates last received a packet, stun packets excluded.
RTCIceCandidateStats.lastPacketSentTimestamp - Web APIs
the rtcicecandidatepairstats property lastpacketsenttimestamp indicates the time at which the connection described by the candidate pair last sent a packet, not including stun packets.
... syntax lastpacketsenttimestamp = rtcicecandidatepairstats.lastpacketsenttimestamp; value a domhighrestimestamp object indicating the timestamp at which the connection described by pair of candidates last sent a packet, stun packets excluded.
RTCIceCandidatePairStats.priority - Web APIs
the higher the value, the more likely the webrtc layer is to select the candidate pair when the time comes to establish (or re-establish) a connection between the two peers.
... syntax pairpriority = rtcicecandidatepairstats.priority; value an integer value indicating the priority of this pair of candidates as compared to other pairs on the same peer connection.
RTCIceCandidatePairStats.requestsReceived - Web APIs
the rtcicecandidatepairstats dictionary's requestsreceived property indicates the total number of stun connectivity check requests that have been received so far on the connection described by this pairing of candidates.
... syntax requestsreceived = rtcicecandidatepairstats.requestsreceived; value an integer value which specifies the number of stun connectivity and/or consent requests that have been received to date on the connection described by this pair of ice candidates.
RTCIceCandidatePairStats.requestsSent - Web APIs
the rtcicecandidatepairstats dictionary's requestssent property indicates the total number of stun connectivity check requests that have been sent so far on the connection described by this pair of candidates.
... syntax requestssent = rtcicecandidatepairstats.requestssent; value an integer value which specifies the number of stun connectivity requests that have been sent to date on the connection described by this pair of ice candidates.
RTCIceCandidatePairStats.responsesReceived - Web APIs
the responsesreceived property in the rtcicecandidatepairstats dictionary indicates the total number of stun connectivity check responses that have been received on the connection described by this pair of candidates.
... syntax responsesreceived = rtcicecandidatepairstats.responsesreceived; value an integer value which specifies the number of stun connectivity request responses that have been received on the connection described by this pair of candidates so far.
RTCIceCandidatePairStats.retransmissionsReceived - Web APIs
the number of retransmissions that have been sent on the connection can be found in retransmissionssent.
... syntax retransmissionsreceived = rtcicecandidatepairstats.retransmissionsreceived; value an integer value indicating the total number of retransmitted stun connectivity check requests have been received on the connection referenced by this candidate pair so far.
RTCIceCandidatePairStats.retransmissionsSent - Web APIs
the number of retransmissions that have been received on the connection can be found in retransmissionsreceived.
... syntax retransmissionssent = rtcicecandidatepairstats.retransmissionssent; value an integer value indicating the total number of retransmitted stun connectivity check requests have been sent on the connection referenced by this candidate pair so far.
RTCIceCandidatePairStats.writable - Web APIs
the obsolete rtcicecandidatepairstats property writable reports whether or not the connection described by the candidate pair is writable.
... syntax iswritable = rtcicecandidatepairstats.writable; value a boolean value which is true if the connection described by this candidate pair has received acknowledgement of receipt (ack) for at least one ice request and that stun consent hasn't expired.
RTCIceCandidateStats - Web APIs
this url matches the one included in the rtcpeerconnectioniceevent object representing the icecandidate event that delivered the candidate to the local peer.
... const isusablenetworktype = stats => { switch(stats.networktype) { case "ethernet": case "vpn": return true; case "bluetooth": case "cellular": case "wimax": case "unknown": default: return false; } } if (rtcstats && rtcstats.type === "local-candidate") { if (!isusablenetworktype(rtcstats)) { abortconnection(); return; } } this code calls a function called abortconnection() if the rtcstats object represents information about a local candidate is which would be using a network connection other than ethernet or a vpn.
RTCIceServer.credential - Web APIs
}; var credential = iceserver.credential; iceserver.credential = newcredential; example this example creates a new rtcpeerconnection which uses a turn server at turnserver.example.org to negotiate connections.
... mypeerconnection = new rtcpeerconnection({ iceservers: [ { urls: "turn:turnserver.example.org", // a turn server username: "webrtc", credential: "turnpassword" } ] }); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtciceserver.credential' in that specification.
RTCIceServer.credentialType - Web APIs
example this example creates a new rtcpeerconnection which will use a turn server at turnserver.example.org to negotiate connections.
... mypeerconnection = new rtcpeerconnection({ iceservers: [ { urls: "turn:turnserver.example.org", // a turn server username: "webrtc", credential: "turnpassword", credentialtype: "password" } ] }); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtciceserver.credential' in that specification.
RTCIceServer.username - Web APIs
}; var username = iceserver.username; iceserver.username = newusername; example this example creates a new rtcpeerconnection which will use a turn server at turnserver.example.org to negotiate connections.
... mypeerconnection = new rtcpeerconnection({ iceservers: [ { urls: "turn:turnserver.example.org", // a turn server username: "webrtc", credential: "turnpassword" } ] }); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtciceserver.username' in that specification.
RTCIceTransport: gatheringstatechange event - Web APIs
bubbles no cancelable no interface event event handler property ongatheringstatechange the key difference between gatheringstatechange and icegatheringstatechange is that the latter represents the overall state of the connection including every rtcicetransport used by every rtcrtpsender and every rtcrtpreceiver on the entire connection.
... examples this example creates a handler for gatheringstatechange events on each rtcrtpsender associated with a given rtcpeerconnection.
RTCIceTransport.getRemoteCandidates() - Web APIs
each time your signaling code calls rtcpeerconnection.addicecandidate() to add a received candidate to the ice session, the ice agent places it in the list returned by this function.
... example this simple example gets the remote candidate list from the rtcicetransport for the first rtcrtpsender on the rtcpeerconnection, then outputs to the console all of the candidates in the list.
RTCIdentityErrorEvent.idp - Web APIs
firefox implements the interface of this property under the following name: rtcpeerconnectionidentityerrorevent.
... it is likely that it will correct this name when it will unprefix rtcpeerconnection, once spec and implementation will have been stabilized.
RTCIdentityErrorEvent.loginUrl - Web APIs
firefox implements the interface of this property under the following name: rtcpeerconnectionidentityerrorevent.
... it is likely that it will correct this name when it will unprefix rtcpeerconnection, once spec and implementation will have been stabilized.
RTCIdentityErrorEvent.protocol - Web APIs
firefox implements the interface of this property under the following name: rtcpeerconnectionidentityerrorevent.
... it is likely that it will correct this name when it will unprefix rtcpeerconnection, once spec and implementation will have been stabilized.
RTCIdentityEvent.assertion - Web APIs
firefox implements the interface this property belongs to under the following name: rtcpeerconnectionidentityevent.
... it is likely that it will correct this name when it will unprefix rtcpeerconnection, once spec and implementation will have been stabilized.
RTCInboundRtpStreamStats.packetsFailedDecryption - Web APIs
the packetsfaileddecryption property of the rtcinboundrtpstreamstats dictionary indicates the total number of rtp packets which failed to be decrypted successfully after being received by the local end of the connection during this session.
... syntax var packetsfaileddecryption = rtcinboundrtpstreamstats.packetsfaileddecryption; value an integer value which indicates how many packets the local end of the rtp connection could not be successfully decrypted.
RTCOfferAnswerOptions.voiceActivityDetection - Web APIs
the voiceactivitydetection property of the rtcofferansweroptions dictionary is used to specify whether or not to use automatic voice detection for the audio on an rtcpeerconnection.
... syntax var options = { voiceactivitydetection: trueorfalse }; value a boolean value indicating whether or not the connection should use voice detection once running.
RTCOfferOptions - Web APIs
the rtcofferoptions dictionary is used to provide optional settings when creating an rtcpeerconnection offer with the createoffer() method.
... icerestart optional a boolean which, when set to true, tells createoffer() to generate and use new values for the identifying properties of the sdp it creates, resulting in a request that triggers renegotiation of the ice connection.
RTCRtpContributingSource.source - Web APIs
oidopera for androidsafari on iossamsung internetsourcechrome full support 59edge full support ≤79firefox full support 59disabled full support 59disabled disabled from version 59: this feature is behind the media.peerconnection.rtpsourcesapi.enable preference (needs to be set to true).
... webview android full support 59chrome android full support 59firefox android full support 59disabled full support 59disabled disabled from version 59: this feature is behind the media.peerconnection.rtpsourcesapi.enable preference (needs to be set to true).
RTCRtpReceiver.getCapabilities() static function - Web APIs
return value an rtcrtpcapabilities object stating what capabilities the browser has for receiving the specified media kind over an rtcpeerconnection.
... example the function below returns a boolean indicating whether or not the device supports receiving h.264 video on a webrtc connection.
RTCRtpStreamStats - Web APIs
the rtcrtpstreamstats dictionary is returned by the rtcpeerconnection.getstats(), rtcrtpsender.getstats(), and rtcrtpreceiver.getstats() methods to provide detailed statistics about webrtc connectivity.
...their primary purpose is to examine the error resiliency of the connection, as they provide information about lost packets, lost frames, and how heavily compressed the data is.
RTCRtpTransceiver.currentDirection - Web APIs
value rtcrtpsender behavior rtcrtpreceiver behavior "sendrecv" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... "sendonly" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
RTCRtpTransceiverDirection - Web APIs
value rtcrtpsender behavior rtcrtpreceiver behavior "sendrecv" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... "sendonly" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
RTCRtpTransceiverInit - Web APIs
the rtcrtptransceiverinit dictionary is used when calling the webrtc function rtcpeerconnection.addtransceiver() to provide configuration options for the new transceiver.
... streams optional a list of mediastream objects to add to the transceiver'srtcrtpreceiver; when the remote peer's rtcpeerconnection's track event occurs, these are the streams that will be specified by that event.
RTCSessionDescription() - Web APIs
this constructor has been deprecated because rtcpeerconnection.setlocaldescription() and other methods which take sdp as input now directly accept an object conforming to the rtcsessiondescriptioninit dictionary, so you don't have to instantiate an rtcsessiondescription yourself.
... this is no longer necessary, however; rtcpeerconnection.setlocaldescription() and other methods which take sdp as input now directly accept an object conforming to the rtcsessiondescriptioninit dictionary, so you don't have to instantiate an rtcsessiondescription yourself.
RTCTrackEvent.track - Web APIs
the webrtc api interface rtctrackevent's read-only track property specifies the mediastreamtrack that has been added to the rtcpeerconnection.
... syntax var track = trackevent.track; value a mediastreamtrack indicating the track which has been added to the rtcpeerconnection.
Resource Timing API - Web APIs
if the resource is loaded via a secure connection a secureconnectionstart timestamp will be available between the connection start and end events.
... the properties which are returned as 0 by default when loading a resource from a domain other than the one of the web page itself: redirectstart, redirectend, domainlookupstart, domainlookupend, connectstart, connectend, secureconnectionstart, requeststart, and responsestart.
SharedWorkerGlobalScope.onconnect - Web APIs
the onconnect property of the sharedworkerglobalscope interface is an event handler representing the code to be called when the connect event is raised — that is, when a messageport connection is opened between the associated sharedworker and the main thread.
...}; examples this example shows a shared worker file — when a connection to the worker occurs from a main thread via a messageport, the onconnect event handler fires.
WebSocket: error event - Web APIs
the error event is fired when a connection with a websocket has been closed due to an error (some data couldn't be sent for example).
... bubbles no cancelable no interface event event handler property onerror examples // create websocket connection const socket = new websocket('ws://localhost:8080'); // listen for possible errors socket.addeventlistener('error', function (event) { console.log('websocket error: ', event); }); specifications specification status html living standardthe definition of 'websocket error' in that specification.
Functions and classes available to Web Workers - Web APIs
10.0 (yes) 10.1 network information api provides information about the system's connection in terms of general connection type (e.g., 'wifi', 'cellular', etc.).
... 28 (28) (yes) (yes) (yes) server-sent events allows a server to push data to a web page at any point, after a connection has been opened to it.
Web APIs
WebAPI
tallevent installtrigger intersectionobserver intersectionobserverentry interventionreportbody k keyboard keyboardevent keyboardlayoutmap keyframeeffect keyframeeffectoptions l largestcontentfulpaint layoutshift layoutshiftattribution linearaccelerationsensor linkstyle localfilesystem localfilesystemsync localmediastream location lock lockmanager lockedfile m midiaccess midiconnectionevent midiinput midiinputmap midimessageevent midioutputmap mscandidatewindowhide mscandidatewindowshow mscandidatewindowupdate msgestureevent msgraphicstrust msmanipulationevent msrangecollection mssitemodeevent magnetometer mathmlelement mediacapabilities mediacapabilitiesinfo mediaconfiguration mediadecodingconfiguration mediadeviceinfo mediadevices mediaelementaudiosourcenode media...
...rstats rtcicecandidatestats rtcicecandidatetype rtcicecomponent rtcicecredentialtype rtcicegathererstate rtciceparameters rtciceprotocol rtcicerole rtciceserver rtcicetcpcandidatetype rtcicetransport rtcicetransportstate rtcidentityassertion rtcidentityerrorevent rtcidentityevent rtcinboundrtpstreamstats rtcnetworktype rtcofferansweroptions rtcofferoptions rtcoutboundrtpstreamstats rtcpeerconnection rtcpeerconnectioniceerrorevent rtcpeerconnectioniceevent rtcremoteoutboundrtpstreamstats rtcrtcpparameters rtcrtpcapabilities rtcrtpcodeccapability rtcrtpcodecparameters rtcrtpcontributingsource rtcrtpencodingparameters rtcrtpparameters rtcrtpreceiveparameters rtcrtpreceiver rtcrtpsendparameters rtcrtpsender rtcrtpstreamstats rtcrtpsynchronizationsource rtcrtptransceiver rtcrtptransceiverdirect...
HTML5 - Developer guides
WebGuideHTMLHTML5
connectivity web sockets allows creating a permanent connection between the page and the server and to exchange non-html data through that means.
... online and offline events firefox 3 supports whatwg online and offline events, which let applications and extensions detect whether or not there's an active internet connection, as well as to detect when the connection goes up and down.
Basics of HTTP - HTTP
connection management in http/1.x http/1.1 was the first version of http to support persistent connection and pipelining.
... connection management in http/2 http/2 completely revisited how connections are created and maintained.
Compression in HTTP - HTTP
compression happens at three different levels: first some file formats are compressed with specific optimized methods, then general encryption can happen at the http level (the resource is transmitted compressed from end to end), and finally compression can be defined at the connection level, between two nodes of an http connection.
...connections between successive intermediate nodes may apply a different compression.
Content-Security-Policy - HTTP
even though the second policy would allow the connection, the first policy contains connect-src 'none'.
... adding additional policies can only further restrict the capabilities of the protected resource, which means that there will be no connection allowed and, as the strictest policy, connect-src 'none' is enforced.
Expect-CT - HTTP
browsers ignore the expect-ct header over http; the header only has effect on https connections.
... enforce optional signals to the user agent that compliance with the certificate transparency policy should be enforced (rather than only reporting compliance) and that the user agent should refuse future connections that violate its certificate transparency policy.
Transfer-Encoding - HTTP
each segment of a multi-node connection can use different transfer-encoding values.
... if you want to compress data over the whole connection, use the end-to-end content-encoding header instead.
OPTIONS - HTTP
WebHTTPMethodsOPTIONS
options /resources/post-here/ http/1.1 host: bar.example accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 accept-language: en-us,en;q=0.5 accept-encoding: gzip,deflate connection: keep-alive origin: https://foo.example access-control-request-method: post access-control-request-headers: x-pingother, content-type the server now can respond if it will accept a request under these circumstances.
... http/1.1 204 no content date: mon, 01 dec 2008 01:15:39 gmt server: apache/2.0.61 (unix) access-control-allow-origin: https://foo.example access-control-allow-methods: post, get, options access-control-allow-headers: x-pingother, content-type access-control-max-age: 86400 vary: accept-encoding, origin keep-alive: timeout=2, max=100 connection: keep-alive specifications specification title rfc 7231, section 4.3.7: options hypertext transfer protocol (http/1.1): semantics and content ...
Network Error Logging - HTTP
tcp.timed_out tcp connection to the server timed out tcp.closed the tcp connection was closed by the server tcp.reset the tcp connection was reset tcp.refused the tcp connection was refused by the server tcp.aborted the tcp connection was aborted tcp.address_invalid the ip address is invalid tcp.address_unreachable the ip address is unreachable tcp.failed the tcp connection failed due to reasons not covered by ...
...previous errors http.error the user agent successfully received a response, but it had a 4xx or 5xx status code http.protocol.error the connection was aborted due to an http protocol error http.response.invalid response is empty, has a content-length mismatch, has improper encoding, and/or other conditions that prevent user agent from processing the response http.response.redirect_loop the request was aborted due to a detected redirect loop http.failed the connection failed due to errors in http protocol not covered by previous errors specifications specification network error logging ...
HTTP
WebHTTP
http follows a classical client-server model, with a client opening a connection to make a request, then waiting until it receives a response.
... connection management in http/1.x describes the three connection management models available in http/1.x, their strengths, and their weaknesses.
Web audio codec guide - Web media technologies
internet explorer opera safari g.711 support 23[1] 15[1] 22[1] no 43[1] 11[1] container support 3gp rtp / webrtc compatible yes licensing all applicable patents have expired, so g.711 is free to use without restriction [1] g.711 is supported only for webrtc connections.
... g.722 is primarily used with webrtc connections, as it's one of the audio codecs mandated by the webrtc specification.
Performance budgets - Web Performance
in order to begin, you need to first measure the devices and connection speeds where your users are coming from (e.g.
... a ~$200 android device over a 3g connection), using multiple tools.
Web Performance
in terms of performance optimization, it's important to optimize to reduce causes of lacency and to test site performance emulating high latency to optimizer for users with lousy connections.
... glossary terms beacon brotli compression client hints code splitting cssom domain sharding effective connection type first contentful paint first cpu idle first input delay first interactive first meaningful paint first paint http http/2 jank latency lazy load long task lossless compression lossy compression main thread minification network throttling packet page load time page prediction parse perceived performance prefetch prerender quic rail real user monitoring resource timi...
Introduction to progressive web apps - Progressive web apps (PWAs)
network independent, so it works offline or with a poor network connection.
... safe, so the connections between the user, the app, and your server are secured against any third parties trying to get access to sensitive data.
method - SVG: Scalable Vector Graphics
WebSVGAttributemethod
cursive fonts), the connections may not align properly when text is rendered along the path.
...with this approach, connected characters, such as in cursive fonts, will maintain their connections.
Web security
connection security transport security layer (tls) the transport layer security (tls) protocol is the standard for enabling two networked applications or devices to exchange information privately and robustly.
...this secure connection allows clients to be sure that they are connected with the intended server, and to exchange sensitive data.
panel - Archive of obsolete content
security warning pages (the "this connection is untrusted" warning) does not work, so panels which trigger it will be broken (bug 1031554).
Appendix C: Avoiding using eval in Add-ons - Archive of obsolete content
if using an unencrypted, insecure connection, a man-in-the-middle attacker might replace the json with attack code before it arrives at the user.
Connecting to Remote Content - Archive of obsolete content
note: you should always test your connection code to cover edge cases, like when there is no internet connection, or the computer is connected to a local network with no internet access (like at an airport or hotel room).
Intercepting Page Loads - Archive of obsolete content
see "recognizing page load failure" at mozilla.dev.extensions for tips on detecting connection failures and http response codes.
Local Storage - Archive of obsolete content
there are two paths you can take when creating the local database you'll be using for your extension: generate the database file (through mozistorageservice.opendatabase(), all tables (through mozistorageconnection.createtable(), and initial data when your extension starts up for the first time.
Updating addons broken by private browsing changes - Archive of obsolete content
nsidownloadmanager now has activeprivatedownloads, activeprivatedownloadcount, and privatedbconnection members to complement the existing public ones.
Inner-browsing extending the browser navigation paradigm - Archive of obsolete content
on slow speed connections, this process can be slow and frustrating.
MMgc - Archive of obsolete content
class myfinalizedobject : public mmgc::gcfinalizedobject { public: ~myfinalizedobject() { // do finalization behavior, like closing network connections, // freeing unmanaged memory owned by this object, etc.
JavaScript Client API - Archive of obsolete content
please note that usage of the sync apis is governed by a terms of service: by accessing or using the firefox sync apis in connection with the development of your own client software to access the firefox sync services (a “third party client”), you acknowledge that you will need to install and use a local version of the firefox sync server for multiple account testing and that any use of mozilla’s hosted firefox sync services is subject to mozilla’s firefox sync terms of service at https://services.mozilla.com/tos/.
HTTP Class Overview - Archive of obsolete content
nshttphandler implements nsiprotocolhandler manages preferences owns the authentication cache holds references to frequently used services nshttpchannel implements nsihttpchannel talks to the cache initiates http transactions processes http response codes intercepts progress notifications nshttpconnection implements nsistreamlistener & nsistreamprovider talks to the socket transport service feeds data to its transaction object routes progress notifications nshttpconnectioninfo identifies a connection nshttptransaction implements nsirequest encapsulates a http request and response parses incoming data nshttpchunkeddecoder owned by a transaction strips chunked transfer encodi...
Running Tamarin acceptance tests - Archive of obsolete content
it's unreliable, losing its adb connection frequently.
Running Tamarin performance tests - Archive of obsolete content
it's unreliable, losing its adb connection frequently.
The life of an HTML HTTP request - Archive of obsolete content
the channel represents the connection to the server, and is the source of the html data stream.
Binding Attachment and Detachment - Archive of obsolete content
for example, given a binding d1, with an explicit inheritance chain of d1 -> d2 -> d3 if this element is attached to an element using element.style.mozbinding that already has a binding chain of s1 -> s2 -> s3 the resulting binding chain following the addition of the binding is d1 -> d2 -> d3 -> s1 -> s2 -> s3 the inheritance between d3 and s1 is implicit, meaning that there is no connection in the xbl documents between the two bindings.
Multiple Rule Example - Archive of obsolete content
there is no reason for this -- it is only used in this example to show that there is no connection between the variables used in the two rules.
Complete - Archive of obsolete content
a preference setting in defaults/preferences/allcustom.js makes the connection between the description and the properties file, specifying the url: chrome://allcustom/locale/install.properties again, the application's chrome registry converts this url to the correct file for the locale you are using.
Using LDAP XPCOM with XULRunner - Archive of obsolete content
actually, the ldap interfaces are available (for exemple, components.interfaces.nsildapconnection is not null), but the components can't be loaded (for example, components.classes["@mozilla.org/network/ldap-connection;1"] is null).
Gecko Compatibility Handbook - Archive of obsolete content
you need to be connected to the internet to test your site on aol; it isn't possible to access your site on a local machine via your lan, without an internet connection.
Mozilla release FAQ - Archive of obsolete content
as to the first case, there's no connection between the mozilla project and the spam.
2006-12-01 - Archive of obsolete content
peter wilson's reply was to add a method that does the deleting with a native implementation that releases the resources held by the object as seen in this database interface: var mydbase = new pgsqlconnection; mydbase.connect("database"); mydbase.exec("select * from mytable where ..."); // use the result data - (native implementation function) mydbase.close() spidermonkey for server side inquiry about why javascript hasn't caught on for general server-side scripting.
NPP_NewStream - Archive of obsolete content
as an optimization to extract the maximum benefit from existing network connections, the browser continues to read data sequentially out of the stream (as in mode np_normal) until the first npn_requestread call is made.
NPSetWindowCallbackStruct - Archive of obsolete content
pointer to the display structure that represents the browser-server connection.
References - Archive of obsolete content
from web standards project web standards group from web standards group web page development: best practices from apple developer connection mozilla web author faq from henri sivonen making your web page compatible with mozilla from nicolás lichtmaier complete css guide from westciv.com css lessons and tutorials from alsacreations html and css lessons and tutorials from htmldog.com preparing for standard-compliant browsers, part 1 from makiko itoh preparing for standard-compliant browsers, part 2 from makiko itoh javascript best practices lists 15 of t...
Desktop gamepad controls - Game development
first, we need an event listener to listen for the connection of the new device: window.addeventlistener("gamepadconnected", gamepadhandler); it's executed once, so we can create some variables we will need later on for storing the controller info and the pressed buttons: var controller = {}; var buttonspressed = []; function gamepadhandler(e) { controller = e.gamepad; output.innerhtml = "gamepad: " + controller.id; } the second line in the game...
Unconventional controls - Game development
connecting the boards and using them may look like this: var cylon = require('cylon'); cylon.robot({ connections: { arduino: { adaptor: 'firmata', port: '/dev/ttyacm0' } }, devices: { makey: { driver: 'makey-button', pin: 2 } }, work: function(my) { my.makey.on('push', function() { console.log("button pushed!"); }); } }).start(); as the description says: this gpio driver allows you to connect a 10 mohm resistor to a digital pin on your arduino or raspberry pi to control you...
Implementing controls using the Gamepad API - Game development
the first one is fired when the browser detects the connection of a new gamepad while the second one is fired when a gamepad is disconnected (either physically by the user or due to inactivity.) in the demo, the gamepadapi object is used to store everything related to the api: var gamepadapi = { controller: {}, turbo: false, connect: function() {}, disconnect: function() {}, update: function() {}, buttonpressed: function() {}, buttons: [], b...
Visual-js game engine - Game development
with strong connection with webpack physics done with (matter.js) matter ts this part is removed from this project.
ALPN - MDN Web Docs Glossary: Definitions of Web-related terms
application-layer protocol negotiation (alpn) is a tls extension which indicates what application layer protocol is negotiating the encryped connection without requiring additional round trips.
Bandwidth - MDN Web Docs Glossary: Definitions of Web-related terms
bandwidth is the measure of how much information can pass through a data connection in a given amount of time.
Distributed Denial of Service - MDN Web Docs Glossary: Definitions of Web-related terms
the united states computer emergency readiness team (us-cert) defines symptoms of denial-of-service attacks to include: unusually slow network performance (opening files or accessing websites) unavailability of a particular website inability to access any website dramatic increase in the number of spam emails received—(this type of dos attack is considered an email bomb) disconnection of a wireless or wired internet connection longterm denial of access to the web or any internet services learn more general knowledge denial-of-service attack on wikipedia ...
Dynamic programming language - MDN Web Docs Glossary: Definitions of Web-related terms
note that while there is indeed a connection between this dynamic/static property of programming languages and dynamic/static-typing, the two are far from synonymous.
Forbidden header name - MDN Web Docs Glossary: Definitions of Web-related terms
forbidden header names start with proxy- or sec-, or are one of the following names: accept-charset accept-encoding access-control-request-headers access-control-request-method connection content-length cookie cookie2 date dnt expect feature-policy host keep-alive origin proxy- sec- referer te trailer transfer-encoding upgrade via note: the user-agent header is no longer forbidden, as per spec — see forbidden header name list (this was implemented in firefox 43) — it can now be set in a fetch headers object, or via xhr setrequestheader().
General header - MDN Web Docs Glossary: Definitions of Web-related terms
the most common general headers are date, cache-control or connection.
HTTP - MDN Web Docs Glossary: Definitions of Web-related terms
resources using the "http" schema are typically transported over unencrypted connections using the http protocol.
HTTP header - MDN Web Docs Glossary: Definitions of Web-related terms
a basic request with one header: get /example.http http/1.1 host: example.com redirects have mandatory headers (location): 302 found location: /newpage.html a typical set of headers: 304 not modified access-control-allow-origin: * age: 2318192 cache-control: public, max-age=315360000 connection: keep-alive date: mon, 18 jul 2016 16:06:00 gmt server: apache vary: accept-encoding via: 1.1 3dc30c7222755f86e824b93feb8b5b8c.cloudfront.net (cloudfront) x-amz-cf-id: tol0fem6ui4fgldrkjx0vao5hpkkgzulyn2twd2gawltr7vlnjtvzw== x-backend-server: developer6.webapp.scl3.mozilla.com x-cache: hit from cloudfront x-cache-info: cached ...
IMAP - MDN Web Docs Glossary: Definitions of Web-related terms
unlike pop3, imap allows multiple simultaneous connections to one mailbox.
IPv4 - MDN Web Docs Glossary: Definitions of Web-related terms
ipv4 is a connectionless protocol to be used on packet-switched link layer networks (ethernet).
IRC - MDN Web Docs Glossary: Definitions of Web-related terms
irc (internet relay chat) is a worldwide chat system requiring an internet connection and an irc client, which sends and receives messages via the irc server.
ISP - MDN Web Docs Glossary: Definitions of Web-related terms
an isp (internet service provider) sells internet access, and sometimes email, web hosting, and voice over ip, either by a dial-up connection over a phone line (formerly more common), or through a broadband connection such as a cable modem or dsl service.
MitM - MDN Web Docs Glossary: Definitions of Web-related terms
learn more owasp article: man-in-the-middle attack wikipedia: man-in-the-middle attack the public-key-pins header (hpkp) can significantly decrease the risk of mitm by instructing browsers to require a whitelisted certificate for all subsequent connections to that website.
Node (networking) - MDN Web Docs Glossary: Definitions of Web-related terms
in networking, a node is a connection point in the network.
Node - MDN Web Docs Glossary: Definitions of Web-related terms
node (networking) in networking, a node is a connection point in the network.
POP3 - MDN Web Docs Glossary: Definitions of Web-related terms
pop3 (post office protocol) is a very common protocol for getting emails from a mail server over a tcp connection.
Preflight request - MDN Web Docs Glossary: Definitions of Web-related terms
w a delete request, before sending a delete request, by using a preflight request: options /resource/foo access-control-request-method: delete access-control-request-headers: origin, x-requested-with origin: https://foo.bar.org if the server allows it, then it will respond to the preflight request with an access-control-allow-methods response header, which lists delete: http/1.1 204 no content connection: keep-alive access-control-allow-origin: https://foo.bar.org access-control-allow-methods: post, get, options, delete access-control-max-age: 86400 the preflight response can be optionally cached for the requests created in the same url using access-control-max-age header like in the above example.
Request header - MDN Web Docs Glossary: Definitions of Web-related terms
a few request headers after a get request: get /home.html http/1.1 host: developer.mozilla.org user-agent: mozilla/5.0 (macintosh; intel mac os x 10.9; rv:50.0) gecko/20100101 firefox/50.0 accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 accept-language: en-us,en;q=0.5 accept-encoding: gzip, deflate, br referer: https://developer.mozilla.org/testpage.html connection: keep-alive upgrade-insecure-requests: 1 if-modified-since: mon, 18 jul 2016 02:36:04 gmt if-none-match: "c561c68d0ba92bbeb8b0fff2a9199f722e3a621a" cache-control: max-age=0 strictly speaking, the content-length header in this example is not a request header like the others, but an entity header: post /myform.html http/1.1 host: developer.mozilla.org user-agent: mozilla/5.0 (macintosh; intel mac...
Response header - MDN Web Docs Glossary: Definitions of Web-related terms
note that strictly speaking, the content-encoding and content-type headers are entity header: 200 ok access-control-allow-origin: * connection: keep-alive content-encoding: gzip content-type: text/html; charset=utf-8 date: mon, 18 jul 2016 16:06:00 gmt etag: "c561c68d0ba92bbeb8b0f612a9199f722e3a621a" keep-alive: timeout=5, max=997 last-modified: mon, 18 jul 2016 02:36:04 gmt server: apache set-cookie: mykey=myvalue; expires=mon, 17-jul-2017 16:06:00 gmt; max-age=31449600; path=/; secure transfer-encoding: chunked vary: cookie, accept-en...
SCTP - MDN Web Docs Glossary: Definitions of Web-related terms
sctp (stream control transmission protocol) is an ietf standard for a transport protocol which enables the reliable, in-order transmission of messages while offering congestion control, multi-homing, and other features to improve reliability and stability of the connection.
SDP - MDN Web Docs Glossary: Definitions of Web-related terms
sdp (session description protocol) is the standard describing a peer-to-peer connection.
TOFU - MDN Web Docs Glossary: Definitions of Web-related terms
if an identifier is found, the client can establish the connection.
TURN - MDN Web Docs Glossary: Definitions of Web-related terms
turn is used by webrtc to allow any two devices on the internet to enter a peer-to-peer connection.
UDP (User Datagram Protocol) - MDN Web Docs Glossary: Definitions of Web-related terms
udp uses a simple connectionless communication model with a minimum of protocol mechanism.
VoIP - MDN Web Docs Glossary: Definitions of Web-related terms
a high speed internet connection is required for voip.
WebSockets - MDN Web Docs Glossary: Definitions of Web-related terms
websocket is a protocol that allows for a persistent tcp connection between server and client so they can exchange data at any time.
firewall - MDN Web Docs Glossary: Definitions of Web-related terms
for example, it can block incoming connections aimed at a certain port or outgoing connections to a certain ip address.
HTTPS - MDN Web Docs Glossary: Definitions of Web-related terms
this secure connection allows clients to safely exchange sensitive data with a server, such as when performing banking activities or online shopping.
Time to first byte - MDN Web Docs Glossary: Definitions of Web-related terms
this time includes dns lookup and establishing the connection using a tcp handshake and ssl handshake if the request is made over https.
MDN Web Docs Glossary: Definitions of Web-related terms
document environment dom (document object model) domain domain name domain sharding dominator dos attack dtls (datagram transport layer security) dtmf (dual-tone multi-frequency signaling) dynamic programming language dynamic typing e ecma ecmascript effective connection type element empty element encapsulation encryption endianness engine entity entity header event exception expando f fallback alignment falsy favicon fetch directive fetch metadata request header firefox os firewall ...
HTML: A good basis for accessibility - Learn web development
it is also much leaner and smaller in terms of code size, which means easier to maintain code, and less bandwidth for the user to download (particularly prevalent for those on slow connections).
HTML: A good basis for accessibility - Learn web development
it is also much leaner and smaller in terms of code size, which means easier to maintain code, and less bandwidth for the user to download (particularly prevalent for those on slow connections).
Mobile accessibility - Learn web development
in general, small screen devices won't need images that are as large as their desktop counterparts, and they are more likely to be on slow network connections.
Accessible multimedia - Learn web development
as well as giving deaf users access to the information contained in the audio, think about a user with a low bandwidth connection, who would find downloading the audio inconvenient.
Sizing items in CSS - Learn web development
downloading overly large images will cause your site to become slow, and it can cost users more money if they are on a metered connection.
What text editors are available? - Learn web development
obviously, that matters too, but a product's cost has little connection with its quality or capability.
How do you make sure your website works properly? - Learn web development
404 errors many times the error just results just from a typo, but sometimes maybe you either forgot to upload a resource or you lost your network connection while you were uploading your resources.
How does the Internet work? - Learn web development
all modern computers can sustain any of those connections.
How much does it cost to do something on the Web? - Learn web development
on the other hand, you'll need a high-bandwidth connection, such as dsl, cable, or fiber access, if you want a more advanced website with hundreds of files, or if you want to deliver heavy video/audio files directly from your web server.
What is accessibility? - Learn web development
the basics of web accessibility a few necessities for basic web accessibility include: whenever your site needs an image to convey meaning, include text as an alternative for visually-challenged users or those with slow connections.
What software do I need to build a website? - Learn web development
you should make sure your hosting provider allows use of a secure connection, e.g.
Front-end web developer - Learn web development
all you need is a computer that can run modern web browsers, an internet connection, and a willingness to learn.
The web and web standards - Learn web development
it also goes beyond people with disabilities — how about young or old people, people from different cultures, people using mobile devices, or people with unreliable or slow network connections?
Creating hyperlinks - Learn web development
for example: if you're on a low bandwidth connection, click a link, and then a multiple megabyte download starts unexpectedly.
Images in HTML - Learn web development
its value is supposed to be a textual description of the image, for use in situations where the image cannot be seen/displayed or takes a long time to render because of a slow internet connection.
From object to iframe — other embedding technologies - Learn web development
these were considered the height of coolness in the mid to late 90s, and there was evidence that having a webpage split up into smaller chunks like this was better for download speeds — especially noticeable with network connections being so slow back then.
HTML table basics - Learn web development
LearnHTMLTablesBasics
a table allows you to quickly and easily look up values that indicate some kind of connection between different types of data, for example a person and their age, or a day of the week, or the timetable for a local swimming pool.
Fetching data from the server - Learn web development
this may not be such a big issue on a desktop on a broadband connection, but it's a major issue on mobile devices and in developing countries that don't have ubiquitous fast internet service.
Third-party APIs - Learn web development
connect the api to your app first, you'll need to make a connection between the api and your app.
Server-side web frameworks - Learn web development
a full-stack http and websocket client/server implementation with ipv6, tls, sni, idna, http/socks5 proxy, unix domain socket, comet (long polling), keep-alive, connection pooling, timeout, cookie, multipart, and gzip compression support.
Ember app structure and componentization - Learn web development
note: the header.js file (shown as skipped) is for connection to a backing glimmer component class, which we don't need for now, as they are for adding interactivity and state manipulation.
Getting started with Svelte - Learn web development
nevertheless, svelte is particularly appropriate to tackle the following situations: web applications intended for low power devices: applications built with svelte have smaller bundle sizes, which is ideal for devices with slow network connections and limited processing power.
Handling common HTML and CSS problems - Learn web development
resolution is a big issue too — for example, mobile devices are less likely to need big heavy images than desktop computers, and are more likely to have slower internet connections and possibly even expensive data plans that make wasted bandwidth more of a problem.
Android-specific test suites
local unit tests are the fastest to develop and execute, but they can only test java code; there's no connection to the gecko rendering engine.
Debugging JavaScript
(note that before firefox 28, this was labeled "browser debugger" and only the debugger was available, not the whole toolbox.) note that you must accept the incoming connection : (you may disable the pop-up above with the devtools.debugger.prompt-connection set to false in about:config.
Simple Thunderbird build
good internet connection for the initial source download.
Eclipse CDT
out of the box, you may/will get gdb connection timeouts.
Experimental features in Firefox
nightly 60 no developer edition 60 no beta 60 no release 60 no preference name security.insecure_connection_text.enabled for normal browsing mode; security.insecure_connection_text.pbmode.enabled for private browsing mode upgrading mixed display content when enabled, this preference causes firefox to automatically upgrade requests for media content from http to https on secure pages.
mozbrowsererror
possible values are: fatal(crash) unknownprotocolfound filenotfound dnsnotfound connectionfailure netinterrupt nettimeout cspblocked phishingblocked malwareblocked unwantedblocked offline malformeduri redirectloop unknownsockettype netreset notcached isprinting deniedportaccess proxyresolvefailure proxyconnectfailure contentencodingfailure remotexul unsafecontenttype corruptedcontenterror certerror other example var browser = document.
How to get a stacktrace with WinDbg
this may take some time depending on your connection speed; the total size of the mozilla and microsoft symbols download is around 1.4gb.
How to investigate Disconnect failures
an example of such a failure disconnect failures happens when one side is closing the connection and mozmill is unable to send the information over the bridge or when firefox crashes.
Webapps.jsm
p: function(aid, aispreinstalled) updateofflinecacheforapp: function(aid) installpreinstalledapp: function installpreinstalledapp(aid) removeifhttpsduplicate: function(aid) installsystemapps: function() loadandupdateapps: function() updatedatastore: function(aid, aorigin, amanifesturl, amanifest) _registersystemmessagesforentrypoint: function(amanifest, aapp, aentrypoint) _registerinterappconnectionsforentrypoint: function(amanifest, aapp,) _registersystemmessages: function(amanifest, aapp) _registerinterappconnections: function(amanifest, aapp) _createactivitiestoregister: function(amanifest, aapp, aentrypoint, arunupdate) _registeractivitiesforapps: function(aappstoregister, arunupdate) _registeractivities: function(amanifest, aapp, arunupdate) _createactivitiestounregister: function...
SVN for Localizers
also, we use https instead of http because changes can only be pushed back to the mozilla svn repository through a secure connection.
Mozilla Port Blocking
"access to the port number given has been disabled for security reasons." "establishing a connection to an unsafe or otherwise banned port was prohibited" "0x804b0013 (ns_error_port_access_not_allowed)" if your product or web site uses a port which is blocked by mozilla's default port blocking rules, you can either change the port of your service to a unblocked value (recommended if possible) or ask your mozilla users to enable the port.
Mozilla Web Developer FAQ
this document answers questions that web authors ask frequently specifically in connection with firefox and other gecko-based browsers.
Named Shared Memory
shared memory protocol named shared memory functions shared memory protocol using named shared memory functions pr_opensharedmemory creates the shared memory segment, if it does not already exist, or opens a connection with the existing shared memory segment if it already exists.
PRLinger
if polarity is set to pr_true and linger is set to 0 (pr_interval_no_wait), the runtime aborts the connection when it is closed and discards any data remaining in the socket send buffer.
PRSockOption
pr_sockopt_keepalive periodically test whether connection is still alive.
PRSocketOptionData
keep_alive periodically test whether connection is still alive.
PR_InitializeNetAddr
this wildcard value is typically used to establish a socket on which to listen for incoming connection requests.
PR_NewUDPSocket
description udp (user datagram protocol) is a connectionless, unreliable datagram protocol of the tcp/ip protocol suite.
PR OpenUDPSocket
description udp (user datagram protocol) is a connectionless, unreliable datagram protocol of the tcp/ip protocol suite.
PR_QueueJob_Accept
causes a job to be queued when a socket has a pending connection.
PR_Read
the value 0 means end of file is reached or the network connection is closed.
PR_Recv
the value 0 means the network connection is closed.
PR_RecvFrom
the value 0 means the network connection is closed.
PR_Shutdown
shuts down part of a full-duplex connection on a specified socket.
PR_TransmitFile
pr_transmitfile_close_socket indicates that the connection should be closed immediately after successful transfer of the file.
NSS Key Log Format
key logs can be written by nss so that external programs can decrypt tls connections.
NSS 3.12.4 release notes
ove des and sha512 for x86_64 platform bug 433791: win16 support should be deleted from nss bug 449332: secu_parsecommandline does not validate its inputs bug 453735: when using cert9 (sqlite3) db, set or change master password fails bug 463544: warning: passing enum* for an int* argument in pkix_validate.c bug 469588: coverity errors reported for softoken bug 470055: pkix_httpcertstore_findsocketconnection reuses closed socket bug 470070: multiple object leaks reported by tinderbox bug 470479: io timeout during cert fetching makes libpkix abort validation.
NSS 3.14.2 release notes
(https://bugzilla.mozilla.org/show_bug.cgi?id=816853) bug 772144 - basic support for running nss test suites on android devices.this is currently limited to running tests from a linux host machine using an ssh connection.
NSS 3.17.4 release notes
notable changes in nss 3.17.4 bug 1084986: if an ssl/tls connection fails, because client and server don't have any common protocol version enabled, nss has been changed to report error code ssl_error_unsupported_version (instead of reporting ssl_error_no_cypher_overlap).
NSS 3.19.2.4 release notes
security fixes in nss 3.19.2.4 the following security fixes from nss 3.21 have been backported to nss 3.19.2.4: bug 1185033 / cve-2016-1979 - use-after-free during processing of der encoded keys in nss bug 1209546 / cve-2016-1978 - use-after-free in nss during ssl connections in low memory bug 1190248 / cve-2016-1938 - errors in mp_div and mp_exptmod cryptographic functions in nss compatibility nss 3.19.2.4 shared libraries are backward compatible with all older nss 3.x shared libraries.
NSS 3.25.1 release notes
previously, with rare server configurations, an md5 signature algorithm might have been selected for client authentication and caused the client to abort the connection soon after.
NSS 3.26.2 release notes
previously, with rare server configurations, an md5 signature algorithm might have been selected for client authentication and caused the client to abort the connection soon after.
nss tech note4
include these files #include "ssl.h" #include "cert.h" get the handle of the cert associated with an ssl connection certcertificate* cert = ssl_peercertificate(prfiledesc *fd); if ssl client, this will get you the server's cert handle; if ssl server, this will get you the client's cert handle if client auth is enabled certcertificate* cert = ssl_localcertificate(prfiledesc *fd); if ssl client, this will get you the client c...
Overview of NSS
the secure sockets layer (ssl) protocol allows mutual authentication between a client and server and the establishment of an authenticated and encrypted connection.
PKCS11 FAQ
MozillaProjectsNSSPKCS11FAQ
if i have a multipurpose token that supports all required pkcs #11 functions and provides rsa_pkcs and dsa mechanisms but but not aes, des or rc4, will nss use the token for the rsa_pkcs mechanisms and the nss internal pkcs #11 module for aes, des or rc4 when making an ssl connection?
NSPR functions
users call nspr socket i/o functions to read from, write to, and shut down an ssl connection, and to close an nspr file descriptor.
TLS Cipher Suite Discovery
in order to communicate securely, an tls client and tls server must agree on the cryptographic algorithms and keys that they will both use on the secured connection.
NSS Tools
the ssltap tool does not decrypt data, but it shows things like the type of ssl message (clienthello, serverhello, etc) and connection data (protocol version, cipher suite, etc).
Multithreading in Necko
socket transport thread (1) socket connections are processed on a single background thread.
Proxies in Necko
this proxy info will only be used if it specifies a socks proxy, through which the connection is then made.
Personal Security Manager (PSM)
these operations include setting up an ssl connection, object signing and signature verification, certificate management (including issuance and revocation), and other common pki functions.
Rhino serialization
serialization provides a way to save the state of an object and write it out to a file or send it across a network connection.
Rhino shell
readurl(url [, charactercoding]) open an input connection to the given string url, read all its bytes and convert them to a string using the specified character coding or default character coding if explicit coding argument is not given.
Getting SpiderMonkey source code
that page also contains links to several bundles, which can be useful if you have a poor network connection.
JS_THREADSAFE
(this is analogous to a database connection pool.) the application has a jscontext that it needs to use each time some event happens.
WebReplayRoadmap
while cloud integration is not in placer, web replay's architecture has recently been redesigned in preparation for cloud support, and communicating with replaying processes over a network connection instead of ipc should work pretty efficiently.
Mozilla Projects
these operations include setting up an ssl connection, object signing and signature verification, certificate management (including issuance and revocation), and other common pki functions.
Pinning violation reports
if a site makes use of key pinning, and your browser sees a certificate chain for that site which does not match the pin, firefox will reject the connection and display an error page.
Browser security
es which are exploitable, and to fix exploitable crashes.handling mozilla security bugsthis document describes how the new security organizational structure will work, and how security-related mozilla bug reports will be handled.pinning violation reportsif a site makes use of key pinning, and your browser sees a certificate chain for that site which does not match the pin, firefox will reject the connection and display an error page.
Using the Places history service
nsiautocompletesearch: url-bar autocomplete from history from 1.9.1 (firefox3.1) on, don't use any places service on (or after) quit-application has been notified, since the database connection will be closed to allow the last sync, and changes will most likely be lost.
XML Extras
you cannot open a connection to a different host.
Using XPCOM Components
components is the javascript object that controls the connection to components, and classes is an array of all of the classes you can ask for by contract id.
XPCOM Stream Guide
MozillaTechXPCOMGuideStreams
(this is particularly true of streams that reach out over a network connection, like http and ftp channels.) stream listeners implement three methods.
mozIStoragePendingStatement
see also storage mozistorageconnection mozistoragestatement ...
mozIStorageResultSet
see also storage mozistorageconnection mozistoragestatement ...
mozIStorageRow
see also storage mozistorageresultset mozistorageconnection mozistoragestatement ...
mozIStorageStatementWrapper
when you call the mozistorageconnection interface's createstatement() method, you get a mozistoragestatement which has just direct bindings to sqlite.
mozIStorageValueArray
see also storage introduction and how-to article mozistorageconnection database connection to a specific file or in-memory data storage mozistoragestatement create and execute sql statements on a sqlite database.
nsIContentPrefService
me); void removegroupedprefs(); void removeobserver(in astring aname, in nsicontentprefobserver aobserver); void removepref(in nsivariant agroup, in astring aname); void removeprefsbyname(in astring aname); void setpref(in nsivariant agroup, in astring aname, in nsivariant avalue); attributes attribute type description dbconnection mozistorageconnection the database connection to the content preferences database.
nsIDOMFile
this allows the file reference to be saved when the form is submitted while the user is using a web application offline, so that the data can be retrieved and uploaded once the internet connection is restored.
nsIDOMMozNetworkStats
to create an instance, use: var networkstatsmanager = components.classes["@mozilla.org/networkstats;1"] .createinstance(components.interfaces.nsidomnetworkstats); attributes attribute type description connectiontype domstring connection type of the stats.
nsIDOMMozNetworkStatsManager
the query may be filtered by connection type and date.
nsIDownloadManager
obsolete since gecko 1.8 dbconnection mozistorageconnection the database connection to the downloads database.
nsIFTPEventSink
last changed in gecko 1.9 (firefox 3) inherits from: nsisupports method overview void onftpcontrollog(in boolean server, in string msg) methods onftpcontrollog allows a consumer to receive a log of the ftp control connection conversation.
nsIFormHistory2
addentry(in astring name, in astring value); boolean entryexists(in astring name, in astring value); boolean nameexists(in astring name); void removeallentries(); void removeentriesbytimeframe(in long long abegintime, in long long aendtime); void removeentriesforname(in astring name); void removeentry(in astring name, in astring value); attributes attribute type description dbconnection mozistorageconnection returns the underlying db connection the form history module is using.
nsIPropertyBag
goodies obtained from window.navigator are: appcodename:"mozilla" appname:"netscape" appversion:"5.0 (windows)" battery:batterymanager buildid:"20140529161749" cookieenabled:true donottrack:"yes" geolocation:geolocation language:"en-us" mimetypes:mimetypearray mozalarms:null mozapps:xpcwrappednative_nohelper mozcameras:cameramanager mozconnection:mozconnection mozcontacts:contactmanager mozid:null mozkeyboard:xpcwrappednative_nohelper mozpay:null mozpermissionsettings:null mozphonenumberservice:phonenumberservice mozpower:mozpowermanager moztcpsocket:null online:true oscpu:"windows nt 5.1" platform:"win32" plugins:pluginarray product:"gecko" productsub:"20100101" useragent:"mozilla/5.0 (windows nt 5.1; rv:30.0) gecko/20100101 firefox/30.0...
nsIProtocolProxyCallback
as with the result of nsiprotocolproxyservice's resolve method, a null result implies that a direct connection should be used.
nsIProtocolProxyFilter
it can also be null to indicate that a direct connection should be used.
nsISocketTransportService
void shutdown(); obsolete since gecko 1.8 attributes attribute type description autodialenabled boolean controls whether or not the socket transport service should poke the autodialer on connection failure.
nsITransport
for a socket transport, these events can include status about the connection.
XPCOM Interface Reference
verimgiencoderimgiloaderimgirequestinidomutilsjsdistackframemoziasyncfaviconsmoziasynchistorymozicoloranalyzermozijssubscriptloadermozipersonaldictionarymoziplaceinfomoziplacesautocompletemoziregistrymozirepresentativecolorcallbackmozispellcheckingenginemozistorageaggregatefunctionmozistorageasyncstatementmozistoragebindingparamsmozistoragebindingparamsarraymozistoragecompletioncallbackmozistorageconnectionmozistorageerrormozistoragefunctionmozistoragependingstatementmozistorageprogresshandlermozistorageresultsetmozistoragerowmozistorageservicemozistoragestatementmozistoragestatementcallbackmozistoragestatementparamsmozistoragestatementrowmozistoragestatementwrappermozistoragevacuumparticipantmozistoragevaluearraymozitxttohtmlconvmozithirdpartyutilmozivisitinfomozivisitinfocallbackmozivisitstatuscal...
Autoconfiguration in Thunderbird
the goal of autoconfiguration is to make it very easy for users to configure the connection of thunderbird to their email servers.
nsIMsgCloudFileProvider
constant value description offlineerr 0x80550014 returned when it appears that there is no active network connection.
Styling the Folder Pane
issecure-{true, false} afolder.server.issecure indicates whether or not the connection to the server will be secure.
Accessibility Inspector - Firefox Developer Tools
this means trying your best to not lock anyone out of accessing information because of any disability they may have, or any other personal circumstances such as the device they are using, the speed of their network connection, or their geographic location or locale.
Add-ons - Firefox Developer Tools
websocket monitor examine the data exchanged in a websocket connection.
Browser Toolbox - Firefox Developer Tools
you will be presented with a dialog like this (it can be removed by setting the devtools.debugger.prompt-connection property to false): click ok, and the browser toolbox will open in its own window: you'll be able to inspect the browser's chrome windows and see, and be able to debug, all the javascript files loaded by the browser itself and by any add-ons that are running.
Dominators - Firefox Developer Tools
each node in the graph represents an object, and each connection between nodes (edge) represents a reference from one object to another.
Dominators view - Firefox Developer Tools
each connection in the graph is labeled with the variable name for the referenced object.
Throttling - Firefox Developer Tools
the network monitor allows you to throttle your network speed to emulate various connection speeds so you can see how your app will behave under different connection types.
Network monitor toolbar - Firefox Developer Tools
throttling menu, to simulate various connection types a menu of other actions: persist logs: by default, the network monitor is cleared each time you navigate to a new page or reload the current page.
Allocations - Firefox Developer Tools
but the main connection between the allocation profile of a site and its responsiveness is the cost of garbage collection (gc).
IndexedDB - Firefox Developer Tools
you can delete an indexeddb database using the context menu in the storage tree: if the database cannot be deleted (most commonly because there are still active connections to the database), a warning message will be displayed in the storage inspector: you can use the context menu in the table widget to delete all items in an object store, or a particular item: ...
Web Console remoting - Firefox Developer Tools
to see how the debugger is used in the web console code, look in browser/devtools/webconsole/webconsole.js, and search for webconsoleconnectionproxy.
about:debugging (before Firefox 68) - Firefox Developer Tools
if you click "debug", you'll see a dialog asking you to accept an incoming connection.
AudioContext.createMediaStreamDestination() - Web APIs
this stream can be used in a similar way as a mediastream obtained via navigator.getusermedia — it can, for example, be sent to a remote peer using the rtcpeerconnection addstream() method.
AudioNode.channelCount - Web APIs
the channelcount property of the audionode interface represents an integer used to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node.
AudioNode - Web APIs
WebAPIAudioNode
audionode.channelcount represents an integer used to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node.
AudioNodeOptions - Web APIs
syntax var audionodeoptions = { "channelcount" : 2, "channelcountmode" : "max", "channelinterpretation" : "discrete" } properties channelcount optional represents an integer used to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node.
BroadcastChannel.close() - Web APIs
the broadcastchannel.close() terminates the connection to the underlying channel, allowing the object to be garbage collected.
Broadcast Channel API - Web APIs
// connection to a broadcast channel const bc = new broadcastchannel('test_channel'); sending a message it is enough to call the postmessage() method on the created broadcastchannel object, which takes any object as an argument.
CloseEvent.initCloseEvent() - Web APIs
wasclean whether or not the connection was cleanly closed.
DataTransfer.effectAllowed - Web APIs
the move operation is used to indicate that the data being dragged will be moved, and the link operation is used to indicate that some form of relationship or connection will be created between the source and drop locations.
Event - Web APIs
WebAPIEvent
rorevent fetchevent focusevent gamepadevent hashchangeevent idbversionchangeevent inputevent keyboardevent mediastreamevent messageevent mouseevent mutationevent offlineaudiocompletionevent overconstrainederror pagetransitionevent paymentrequestupdateevent pointerevent popstateevent progressevent relatedevent rtcdatachannelevent rtcidentityerrorevent rtcidentityevent rtcpeerconnectioniceevent sensorevent storageevent svgevent svgzoomevent timeevent touchevent trackevent transitionevent uievent userproximityevent webglcontextevent wheelevent constructor event() creates an event object, returning it to the caller.
EventSource() - Web APIs
configuration optional provides options to configure the new connection.
EventSource: error event - Web APIs
the error event of the eventsource api is fired when a connection with an event source fails to be opened.
FileHandle API - Web APIs
api overview this api is based on the following interfaces: idbdatabase.mozcreatefilehandle (was called idbdatabase.mozcreatefilehandle.) idbmutablefile (was previously filehandle.) lockedfile filerequest it also has connections with the file api, especially the file and blob interfaces.
HTMLCanvasElement.captureStream() - Web APIs
send it to another computer using an rtcpeerconnection // pc is an rtcpeerconnection created elsewhere pc.addstream(stream); specifications specification status comment media capture from dom elementsthe definition of 'htmlcanvaselement.capturestream()' in that specification.
HTMLMediaElement.captureStream() - Web APIs
this can be used, for example, as a source for a webrtc rtcpeerconnection.
Drag Operations - Web APIs
the move operation is used to indicate that the data being dragged will be moved, and the link operation is used to indicate that some form of relationship or connection will be created between the source and drop locations.
HTML Drag and Drop API - Web APIs
link indicates that some form of relationship or connection will be created between the source and drop locations.
IDBDatabase: abort event - Web APIs
the abort event is fired on idbdatabase when a transaction is aborted and bubbles up to the connection object.
IDBDatabase: error event - Web APIs
the error event is fired on idbdatabase when a request returns an error and the event bubbles up to the connection object.
IDBDatabase.name - Web APIs
WebAPIIDBDatabasename
example this example shows a database connection being opened, the resulting idbdatabase object being stored in a db variable, and the name property then being logged.
IDBDatabase.onabort - Web APIs
the onabort event handler of the idbdatabase interface handles the abort event, fired when a transaction is aborted and bubbles up to the connection object.
IDBDatabase.onclose - Web APIs
the close event is fired after all transactions have been aborted and the connection has been closed.
IDBDatabase.onerror - Web APIs
the onerror event handler of the idbdatabase interface handles the error event, fired when a request returns an error and bubbles up to the connection object.
IDBDatabase.transaction() - Web APIs
example in this example we open a database connection, then use transaction() to open a transaction on the database.
IDBDatabaseSync - Web APIs
the databasesync interface in the indexeddb api represents a synchronous connection to a database.
IDBFactory.deleteDatabase() - Web APIs
when deletedatabase() is called, any other open connections to this particular database will get a versionchange event.
IDBFactory - Web APIs
methods idbfactory.open the current method to request opening a connection to a database.
IDBOpenDBRequest: blocked event - Web APIs
the blocked handler is executed when an open connection to a database is blocking a versionchange transaction on the same database.
IDBOpenDBRequest - Web APIs
blocked fired when an open connection to a database is blocking a versionchange transaction on the same database.
IDBTransaction - Web APIs
properties idbtransaction.db read only the database connection with which this transaction is associated.
IDBTransactionSync - Web APIs
method overview void abort() raises (idbdatabaseexception); void commit() raises (idbdatabaseexception); idbobjectstoresync objectstore(in domstring name) raises (idbdatabaseexception); attributes attribute type description db idbdatabasesync the database connection that this transaction is associated with.
Key Values - Web APIs
keycode_tv_media_context_menu (257) "tvnetwork" toggle the tv's network connection on and off.
MIDIMessageEvent - Web APIs
properties midiconnectionevent.data a uint8array containing the data bytes of a single midi message.
MediaDevices.getDisplayMedia() - Web APIs
the stream is then returned to the caller for use, perhaps for adding to a webrtc call using rtcpeerconnection.addtrack() to add the video track from the stream.
MediaEncodingConfiguration - Web APIs
using rtcpeerconnection).
MediaStream.onaddtrack - Web APIs
this event is fired when the browser adds a track to the stream (such as when a rtcpeerconnection is renegotiated or a stream being captured using htmlmediaelement.capturestream() gets a new set of tracks because the media element being captured loaded a new source.
MediaStream.onremovetrack - Web APIs
this event is fired when the browser removes a track from the stream (such as when a rtcpeerconnection is renegotiated or a stream being captured using htmlmediaelement.capturestream() gets a new set of tracks because the media element being captured loaded a new source.
MediaStreamAudioSourceNode - Web APIs
this media could be from a microphone (through getusermedia()) or from a remote peer on a webrtc call (using the rtcpeerconnection's audio tracks).
MediaStreamEvent.stream - Web APIs
syntax var stream = event.stream; example pc.onaddstream = function( ev ) { alert("a stream (id: '" + ev.stream.id + "') has been added to this connection."); }; ...
MediaStreamEvent - Web APIs
examples pc.onaddstream = function( ev ) { alert("a stream (id: '" + ev.stream.id + "') has been added to this connection."); }; ...
MediaStreamTrack.onended - Web APIs
this event occurs when the track will no longer provide data to the stream for any reason, including the end of the media input being reached, the user revoking needed permissions, the source device being removed, or the remote peer ending a connection.
MediaStreamTrack.remote - Web APIs
it returns a boolean with a value of true if the track is sourced remotely (that is, sourced by an rtcpeerconnection), or false if it is sourced locally.
MediaStreamTrack - Web APIs
mediastreamtrack.remote read only returns a boolean with a value of true if the track is sourced by a rtcpeerconnection, false otherwise.
MediaStreamTrackAudioSourceNode - Web APIs
the audio itself might be input from a microphone or other audio sampling device, or might be received through a rtcpeerconnection, among other posible options.
MediaTrackConstraints.deviceId - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackConstraints.echoCancellation - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackConstraints.facingMode - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackConstraints.latency - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackConstraints - Web APIs
for example, because rtp doesn't provide some of these values during negotiation of a webrtc connection, a track associated with a rtcpeerconnection will not include certain values, such as facingmode or groupid.
MediaTrackSettings.deviceId - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackSettings.facingMode - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackSettings.groupId - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackSettings.latency - Web APIs
because rtp doesn't include this information, tracks associated with a webrtc rtcpeerconnection will never include this property.
MediaTrackSettings - Web APIs
for example, because rtp doesn't provide some of these values during negotiation of a webrtc connection, a track associated with a rtcpeerconnection will not include certain values, such as facingmode or groupid.
Using the Media Capabilities API - Web APIs
a mediarecorder object) or transmission (for media transmitted over electronic means like rtcpeerconnection) — plus either an audio or video configuration as described above.
Media Source API - Web APIs
it can however support significantly more client connections than webrtc.
NDEFReader.scan() - Web APIs
WebAPINDEFReaderscan
notsupportederror there is no nfc adapter compatible with web nfc, or a connection can not be established.
NDEFWriter.write() - Web APIs
WebAPINDEFWriterwrite
notsupportederror there is no nfc adapter compatible with web nfc, or the available nfc adapter does not support pushing messages, or connection can not be established.
Navigator - Web APIs
WebAPINavigator
navigator.connection read only provides a networkinformation object containing information about the network connection of a device.
NetworkInformation.effectiveType - Web APIs
the effectivetype read-only property of the networkinformation interface returns the effective type of the connection meaning one of 'slow-2g', '2g', '3g', or '4g'.
NetworkInformation.type - Web APIs
the networkinformation.type read-only property returns the type of connection a device is using to communicate with the network.
OfflineAudioContext.startRendering() - Web APIs
the startrendering() method of the offlineaudiocontext interface starts rendering the audio graph, taking into account the current connections and the current scheduled changes.
OfflineAudioContext - Web APIs
offlineaudiocontext.startrendering() starts rendering the audio, taking into account the current connections and the current scheduled changes.
Page Visibility API - Web APIs
tabs running code that's using real-time network connections (websockets and webrtc) go unthrottled in order to avoid closing these connections timing out and getting unexpectedly closed.
PerformanceResourceTiming.domainLookupEnd - Web APIs
; } } function print_start_and_end_properties(perfentry) { // print timestamps of the *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.domainLookupStart - Web APIs
print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.fetchStart - Web APIs
print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.nextHopProtocol - Web APIs
when a proxy is used, if a tunnel connection is established, this property returns the alpn protocol id of the tunneled protocol.
PerformanceResourceTiming.redirectEnd - Web APIs
print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.redirectStart - Web APIs
print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.responseStart - Web APIs
print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceResourceTiming.workerStart - Web APIs
print_start_and_end_properties(perfentry) { // print timestamps of the performanceentry *start and *end properties properties = ["connectstart", "connectend", "domainlookupstart", "domainlookupend", "fetchstart", "redirectstart", "redirectend", "requeststart", "responsestart", "responseend", "secureconnectionstart", "workerstart"]; for (var i=0; i < properties.length; i++) { // check each property var supported = properties[i] in perfentry; if (supported) { var value = perfentry[properties[i]]; console.log("...
PerformanceTiming.domainLookupEnd - Web APIs
if a persistent connection is used, or the information is stored in a cache or a local resource, the value will be the same as performancetiming.fetchstart.
PerformanceTiming.domainLookupStart - Web APIs
if a persistent connection is used, or the information is stored in a cache or a local resource, the value will be the same as performancetiming.fetchstart.
PerformanceTiming.requestStart - Web APIs
if the transport layer fails after the start of the request and the connection is reopened, this property will be set to the time corresponding to the new request.
PerformanceTiming.responseEnd - Web APIs
the legacy performancetiming.responseend read-only property returns an unsigned long long representing the moment, in miliseconds since the unix epoch, when the browser received the last byte of the response, or when the connection is closed if this happened first, from the server from a cache or from a local resource.
RTCAnswerOptions - Web APIs
the rtcansweroptions dictionary is used to provide optional settings when creating an sdp answer using rtcpeerconnection.createoffer() after receiving an offer from a remote peer.
RTCCertificate - Web APIs
the interface of the the webrtc api provides an object represents a certificate that an rtcpeerconnection uses to authenticate.
RTCDTMFSender.toneBuffer - Web APIs
the rtcdtmfsender interface's tonebuffer property returns a string containing a list of the dtmf tones currently queued for sending to the remote peer over the rtcpeerconnection.
RTCDataChannel.binaryType - Web APIs
var dc = peerconnection.createdatachannel("binary"); dc.binarytype = "arraybuffer"; dc.onmessage = function(event) { let bytearray = new uint8array(event.data); let hexstring = ""; bytearray.foreach(function(byte) { hexstring += byte.tostring(16) + " "; }); }; specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rt...
RTCDataChannel.bufferedAmount - Web APIs
var dc = peerconnection.createdatachannel("file transfer"); /* ...
RTCDataChannel.bufferedAmountLowThreshold - Web APIs
var dc = peerconnection.createdatachannel("file transfer"); dc.bufferedamountlowthreshold = 65535; dc.onbufferedamountlow = function() { /* use send() to queue more data to be sent */ }; specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.bufferedamountlowthreshold' in that specification.
RTCDataChannel: bufferedamountlow event - Web APIs
let pc = new rtcpeerconnection(); let dc = pc.createdatachannel("sendfile"); let source = /* source data object */ dc.bufferedamountlowthreshold = 65536; pc.addeventlistener("bufferedamountlow", ev => { if (source.position <= source.length) { dc.send(source.readfile(65536)); } }, false); after creating the rtcpeerconnection, this calls rtcpeerconnection.createdatachannel() to create the data channel.
RTCDataChannel.maxPacketLifeTime - Web APIs
if not set when rtcpeerconnection.createdatachannel() was called to create the data channel, this value is null.
RTCDataChannel: message event - Web APIs
examples for a given rtcdatachannel, dc, created for a peer connection using its createdatachannel() method, this code sets up a handler for incoming messages and acts on them by adding the data contained within the message to the current document as a new <p> (paragraph) element.
RTCDataChannel.onbufferedamountlow - Web APIs
pc = new rtcpeerconnection(); dc = pc.createdatachannel("sendfile"); /* ...
RTCDataChannel.onerror - Web APIs
let pc = new rtcpeerconnection(); let dc = pc.createdatachannel("playercontrol"); dc.onerror = function(event) { myuilibrary.doalertbox({ 'network error', // alert title 'the error "' + event.message + '" occurred while handling player control network messages.', event.filename, event.lineno, event.colno }); } /* ...
RTCDataChannel: open event - Web APIs
bubbles no cancelable no interface rtcdatachannelevent event handler property onopen examples this example adds to the rtcdatachannel dc a handler for the open event that adjusts the user interface to indicate that a chat window is ready to be used after a connection has been established.
RTCDataChannel.protocol - Web APIs
example var pc = new rtcpeerconnection(); var dc = pc.createdatachannel("my channel", { protocol: "json" }); function handlechannelmessage(datachannel, msg) { switch(datachannel.protocol) { case "json": /* process json data */ break; case "raw": /* process raw binary data */ break; } } specifications specification status comment webrtc 1.0: real-time communi...
RTCDataChannel.reliable - Web APIs
syntax var reliable = adatachannel.reliable; value true if the rtcdatachannel's connection is reliable; false if it isn't.
RTCDataChannelEvent.channel - Web APIs
syntax var channel = rtcdatachannelevent.channel; value a rtcdatachannel object representing the data channel linking the receiving rtcpeerconnection to its remote peer.
RTCErrorEvent - Web APIs
the most common of these is probably rtcpeerconnectioniceerrorevent, used by the icecandidateerror event, which signals an error that has occurred while gathering ice candidates during connection negotiation.
RTCIceCandidate.address - Web APIs
security notes it's important to note here that although webrtc does not require the two peers on an rtcpeerconnection to know one another's true ip addresses, the address property on rtcicecandidate can expose more information about the source of the remote peer than the user expects.
RTCIceCandidate.priority - Web APIs
the a-line string is obtained either from the rtcicecandidateinit property candidate or from an a-line string passed into rtcpeerconnection.addicecandidate() instead of an rtcicecandidate.
RTCIceCandidate.protocol - Web APIs
protocol is null by default if not specified properly in the sdp, but this is an error condition and will result in a thrown exception when you call rtcpeerconnection.addicecandidate().
RTCIceCandidate.type - Web APIs
if type is null, that information was missing from the candidate's a-line, which will cause rtcpeerconnection.addicecandidate() to throw an operationerror exception.
RTCIceCandidateInit - Web APIs
it's also used as the return value from the rtcicecandidate.tojson() method, and can be passed directly into rtcpeerconnection.addicecandidate() to add a candidate to the peer connection.
RTCIceCandidatePairStats.consentExpiredTimestamp - Web APIs
this indicates when the current stun bindings — the mapping of the ip address and port configurations for both peers on the webrtc connection — are due to expire.
RTCIceCandidatePairStats.firstRequestTimestamp - Web APIs
syntax firstrequesttimestamp = rtcicecandidatepairstats.firstrequesttimestamp; value a domhighrestimestamp object indicating the timestamp at which the first stun request was sent on the connection described by the described pair of candidates.
RTCIceCandidatePairStats.lastRequestTimestamp - Web APIs
syntax lastrequesttimestamp = rtcicecandidatepairstats.lastrequesttimestamp; value a domhighrestimestamp object indicating the timestamp at which the last (most recent) stun request was sent on the connection indicated by the described pair of candidates.
RTCIceCandidateStats.lastResponseTimestamp - Web APIs
syntax lastresponsetimestamp = rtcicecandidatepairstats.lastresponsetimestamp; value a domhighrestimestamp object indicating the timestamp at which the most recent stun response was received on the connection defined by the described pair of candidates.
RTCIceCandidateStats.localCandidateId - Web APIs
syntax localcandidateid = rtcicecandidatepairstats.localcandidateid; value a domstring giving a unique identifier for the local rtcicecandidate for the connection described by this rtcicecandidatepairstats object.
RTCIceCandidatePairStats.responsesSent - Web APIs
the rtcicecandidatepairstats dictionary's responsessent property indicates the total number of stun connectivity check responses that have been sent so far on the connection described by this pair of candidates.
RTCIceCandidatePairStats.selected - Web APIs
example the function shown in this example identifies the currently-selected candidate pair from a statistics report by first iterating over each report, looking for a transport report; when one is found, that transport's selectedcandidatepairid is used to get the rtcicecandidatepair describing the connection.
RTCIceCandidatePairStats.totalRoundTripTime - Web APIs
syntax totalrtt = rtcicecandidatepairstats.totalroundtriptime; value this floating-point value indicates the total number of seconds which have elapsed between sending out stun connectivity and consent check requests and receiving their responses, for all such requests made so far on the connection described by this candidate pair.
RTCIceCandidateStats.deleted - Web APIs
window.setinterval(function() { mypeerconnection.getstats(null).then(stats => { let statsoutput = ""; stats.foreach(report => { if ((stats.type === "local-candidate" || stats.type === "remote.candidate") && !stats.deleted) { statsoutput += `<h2>report: ${report.type}</h3>\n<strong>id:</strong> ${report.id}<br>\n` + `<strong>timestamp:</strong> ${report.timestamp}<br>\n`; // now the statisti...
RTCIceCandidateStats.priority - Web APIs
during ice negotiation while setting up a webrtc peer connection, the priority values reported to the remote peer by a user agent are used to determine which candidates are considered "more desirable".
RTCIceParameters.usernameFragment - Web APIs
see rtcicecandidate.usernamefragment to learn more about username fragments and their role in a connection.
RTCIceRole - Web APIs
the rtcicerole enumerated type lists the string values that identify whether a connection's ice agent is serving as the controlling agent or the controlled agent, as indicated by rtcicetransport.role.
RTCIceTransport.getLocalCandidates() - Web APIs
example this simple example gets the local candidate list from the rtcicetransport for the first rtcrtpsender on the rtcpeerconnection, then outputs to the console all of the candidates in the list.
RTCIceTransport.getLocalParameters() - Web APIs
the local peer's parameters are obtained during ice signaling and delivered to the transport when the client calls rtcpeerconnection.setlocaldescription().
RTCIceTransport.getRemoteParameters() - Web APIs
the remote peer's parameters are received during ice signaling and delivered to the transport when the client calls rtcpeerconnection.setremotedescription().
RTCIceTransport.onselectedcandidatepairchange - Web APIs
the rtcicetransport interface's onselectedcandidatepairchange event handler specifies a function to be called to handle the selectedcandidatepairchange event, which is fired when the ice agent selects a new candidate pair to be used for the connection.
RTCIceTransport.onstatechange - Web APIs
example this snippet establishes a handler for the statechange event that looks to see if the transport has entered the "failed" state, which indicates that the connection has failed with no chance of being automatically restored.
RTCIceTransport: statechange event - Web APIs
bubbles no cancelable no interface event event handler property rtcicetransport.onstatechange examples given an rtcpeerconnection, pc, the following code creates an event handler that calls a function named handlefailure() if the ice transport enters a failure state.
RTCIdentityAssertion - Web APIs
the rtcidentityassertion interface of the the webrtc api represents the identity of the a remote peer of the current connection.
RTCInboundRtpStreamStats.firCount - Web APIs
syntax var fircount = rtcinboundrtpstreamstats.fircount; value an integer value indicating how many fir packets have been received by the sender during the current connection.
RTCInboundRtpStreamStats - Web APIs
the webrtc api's rtcinboundrtpstreamstats dictionary, based upon rtcreceivedrtpstreamstats and rtcstats, contains statistics related to the receiving end of an rtp stream on the local end of the rtcpeerconnection.
RTCOfferAnswerOptions - Web APIs
it's used as the base type for the options parameter when calling createoffer() or createanswer() on an rtcpeerconnection.
RTCOutboundRtpStreamStats.firCount - Web APIs
syntax var fircount = rtcoutboundrtpstreamstats.fircount; value an integer value indicating how many fir packets have been received by the sender during the current connection.
RTCRtcpParameters - Web APIs
the rtcrtcpparameters dictionary provides parameters of an rtcp connection.
RTCRtpCapabilities - Web APIs
one possible value is video/ulpfec (a generic error connection model).
RTCRtpCodecParameters - Web APIs
in addition to being the type of the rtcrtpparameters.codecs property, it's used when calling rtcrtptransceiver.setcodecpreferences() to configure a transceiver's codecs before beginning the offer/answer process to establish a webrtc peer connection.
RTCRtpEncodingParameters.scaleResolutionDownBy - Web APIs
therefore, specifying a value less than 1.0 is not permitted and will cause a rangeerror exception to be thrown by rtcpeerconnection.addtransceiver() or rtcrtpsender.setparameters().
RTCRtpReceiveParameters - Web APIs
it describes the parameters being used by the receiver's rtp connection to the remote peer.
RTCRtpReceiver.getStats() - Web APIs
the rtcrtpreceiver method getstats() asynchronously requests an rtcstatsreport object which provides statistics about incoming traffic on the owning rtcpeerconnection, returning a promise whose fulfillment handler will be called once the results are available.
RTCRtpReceiver.transport - Web APIs
note that when bundling is in effect—that is, when the rtcpeerconnection was created with an rtcconfiguration object whose bundlepolicy is max-compat or max-bundle—multiple receivers may be sharing the same transport; in this case, all of them are using the same connection to transmit and/or receive rtp and rtcp packets.
RTCRtpReceiver - Web APIs
the rtcrtpreceiver interface of the webrtc api manages the reception and decoding of data for a mediastreamtrack on an rtcpeerconnection.
RTCRtpSender.getCapabilities() static function - Web APIs
return value an rtcrtpcapabilities object stating what capabilities the browser has for sending the specified media kind over an rtcpeerconnection.
RTCRtpSender.getStats() - Web APIs
the rtcrtpsender method getstats() asynchronously requests an rtcstatsreport object which provides statistics about outgoing traffic on the rtcpeerconnection which owns the sender, returning a promise which is fulfilled when the results are available.
RTCRtpSender.setParameters() - Web APIs
in other words, setparameters() updates the configuration of the rtp transmission as well as the encoding configuration for a specific media track on the webrtc connection.
RTCRtpSender.transport - Web APIs
note that when bundling is in effect—that is, when the rtcpeerconnection was created with an rtcconfiguration object whose bundlepolicy is max-compat or max-bundle—multiple senders may be sharing the same transport; in this case, all of them are using the same connection to transmit and/or receive rtp and rtcp packets.
RTCRtpSender - Web APIs
if null, the track and/or the connection doesn't support dtmf.
RTCRtpStreamStats.firCount - Web APIs
syntax var fircount = rtcrtpstreamstats.fircount; value an integer value indicating how many fir packets have been received by the sender during the current connection.
RTCSessionDescription.sdp - Web APIs
value the value is a domstring containing an sdp message like this one: v=0 o=alice 2890844526 2890844526 in ip4 host.anywhere.com s= c=in ip4 host.anywhere.com t=0 0 m=audio 49170 rtp/avp 0 a=rtpmap:0 pcmu/8000 m=video 51372 rtp/avp 31 a=rtpmap:31 h261/90000 m=video 53000 rtp/avp 32 a=rtpmap:32 mpv/90000 example // the remote description has been set previously on pc, an rtcpeerconnection alert(pc.remotedescription.sdp); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcsessiondescription.sdp' in that specification.
RTCSessionDescription.type - Web APIs
example // the remote description has been set previously on pc, a rtcpeerconnection alert(pc.remotedescription.type); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcsessiondescription.type' in that specification.
RTCSessionDescriptionCallback - Web APIs
example var pc = new rtcpeerconnection(); var descriptioncallback = function(offer) { pc.setlocaldescription(offer); } pc.createoffer(descriptioncallback); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcsessiondescriptioncallback' in that specification.
RTCStats.id - Web APIs
WebAPIRTCStatsid
using the id, you can correlate two or more rtcstats-based objects in order to monitor statistics over time for a given webrtc object, such as an rtp stream, an rtcpeerconnection, or an rtcdatachannel.
RTCStats - Web APIs
WebAPIRTCStats
rtcinboundrtpstreamstats contains statistics that can only be measured on a receiver at the local end of the rtp connection.
RTCStatsIceCandidatePairState - Web APIs
this represents the state of this candidate pair within the ice check list for the rtcpeerconnection.
RTCTrackEvent.streams - Web APIs
the webrtc api interface rtctrackevent's read-only streams property specifies an array of mediastream objects, one for each of the streams that comprise the track being added to the rtcpeerconnection.
RTCTrackEventInit.track - Web APIs
this is the track that's being added to the rtcpeerconnection.
Server-sent events - Web APIs
interfaces eventsource defines all the features that handle connecting to a server, receiving events/data, errors, closing a connection, etc.
Using Service Workers - Web APIs
in the future they will have throttling/offline modes to simulate bad or non-existent connections, which will be a really good thing.
SharedWorkerGlobalScope: connect event - Web APIs
bubbles no cancelable no interface messageevent event handler property sharedworkerglobalscope.onconnect examples this example shows a shared worker file — when a connection to the worker occurs from a main thread via a messageport, the onconnect event handler fires.
SharedWorkerGlobalScope - Web APIs
sharedworkerglobalscope.onconnect is an eventhandler representing the code to be called when the connect event is raised — that is, when a messageport connection is opened between the associated sharedworker and the main thread.
TrackEvent - Web APIs
it's important not to confuse trackevent with the rtctrackevent interface, which is used for tracks which are part of an rtcpeerconnection.
WebGLRenderingContext.isContextLost() - Web APIs
examples include: two or more pages are using the gpu, but together place too high a demand on the gpu, so the browser tells the two contexts that they've lost the connection, then selects one of the two to restore access for.
High-level guides - Web APIs
the high-level guides listed below introduce webrtc technology from a top-down perspective, describing the overall architecture, the life cycle of a webrtc connection, and basic security and technological issues you might run into as you explore and build web content or apps using the webrtc api.
Improving compatibility using WebRTC adapter.js - Web APIs
for example, on firefox versions older than 38, the adapter adds the rtcpeerconnection.urls property; firefox doesn't natively support this property until firefox 38, while on chrome, the adapter adds support for the promise based api is added if it's not present.
WebSocket() - Web APIs
exceptions thrown security_err the port to which the connection is being attempted is being blocked.
WebSocket.binaryType - Web APIs
the websocket.binarytype property returns the type of binary data being transmitted by the connection.
WebSocket.bufferedAmount - Web APIs
this value does not reset to zero when the connection is closed; if you keep calling send(), this will continue to climb.
WebSocket.extensions - Web APIs
this is currently only the empty string or a list of extensions as negotiated by the connection.
WebSocket: message event - Web APIs
bubbles no cancelable no interface messageevent event handler property onmessage examples // create websocket connection.
WebSocket.onclose - Web APIs
WebAPIWebSocketonclose
the websocket.onclose property is an eventhandler that is called when the websocket connection's readystate changes to closed.
WebSocket.onerror - Web APIs
WebAPIWebSocketonerror
syntax websocket.onerror = eventhandler; value a function or eventhandler which is executed whenever an error event occurs on the websocket connection.
WebSocket.onopen - Web APIs
WebAPIWebSocketonopen
the websocket.onopen property is an eventhandler that is called when the websocket connection's readystate changes to 1; this indicates that the connection is ready to send and receive data.
WebSocket.protocol - Web APIs
the websocket.protocol read-only property returns the name of the sub-protocol the server selected; this will be one of the strings specified in the protocols parameter when creating the websocket object, or the empty string if no connection is established.
Basic concepts behind Web Audio API - Web APIs
establish connections from the audio sources through zero or more effects, eventually ending at the chosen destination.
Web Audio API - Web APIs
this last connection is only necessary if the user is supposed to hear the audio.
Using the Web Speech API - Web APIs
ully — the speechrecognitionerror.error property contains the actual error returned: recognition.onerror = function(event) { diagnostic.textcontent = 'error occurred in recognition: ' + event.error; } speech synthesis speech synthesis (aka text-to-speech, or tts) involves receiving synthesising text contained within an app to speech, and playing it out of a device's speaker or audio output connection.
Window: gamepaddisconnected event - Web APIs
bubbles no cancelable no interface gamepadevent event handler property ongamepaddisconnected examples window.addeventlistener('gamepaddisconnected', event => { console.log('lost connection with the gamepad.'); }); specifications specification status gamepad working draft ...
Window: offline event - Web APIs
bubbles no cancelable no interface event event handler property onoffline examples // addeventlistener version window.addeventlistener('offline', (event) => { console.log("the network connection has been lost."); }); // onoffline version window.onoffline = (event) => { console.log("the network connection has been lost."); }; specifications specification status html living standardthe definition of 'offline event' in that specification.
WorkerNavigator - Web APIs
workernavigator.connectionread only provides a networkinformation object containing information about the network connection of a device.
Using XMLHttpRequest - Web APIs
note: starting in gecko 9.0, progress events can now be relied upon to come in for every chunk of data received, including the last chunk in cases in which the last packet is received and the connection closed before the progress event is fired.
XMLHttpRequest() - Web APIs
mozsystem boolean: setting this flag to true allows making cross-site connections without requiring the server to opt-in using cors.
XMLHttpRequest.getAllResponseHeaders() - Web APIs
an example of what a raw header string looks like: date: fri, 08 dec 2017 21:04:30 gmt\r\n content-encoding: gzip\r\n x-content-type-options: nosniff\r\n server: meinheld/0.6.1\r\n x-frame-options: deny\r\n content-type: text/html; charset=utf-8\r\n connection: keep-alive\r\n strict-transport-security: max-age=63072000\r\n vary: cookie, accept-encoding\r\n content-length: 6502\r\n x-xss-protection: 1; mode=block\r\n each line is terminated by both carriage return and line feed characters (\r\n).
XRSession.end() - Web APIs
WebAPIXRSessionend
you can use the promise to do things like update ui elements to reflect the shut down connection, trigger application shut down, or whatever else you might need to do.
Using the alert role - Accessibility
the alert role is most useful for information that requires the user's immediate attention, for example: an invalid value was entered into a form field the user's login session is about to expire the connection to the server was lost, local changes will not be saved because of its intrusive nature, the alert role must be used sparingly and only in situations where the user's immediate attention is required.
@font-face - CSS: Cascading Style Sheets
the ability to specify the name of a locally-installed font to look for and use makes it possible to customize the font beyond the basics while making it possible to do so without relying on an internet connection.
image-set() - CSS: Cascading Style Sheets
WebCSSimage-set
resolution can be used as a proxy for filesize — a user agent on a slow mobile connection with a high-resolution screen may prefer to receive lower-resolution images rather than waiting for a higher resolution image to load.
Guide to scroll anchoring - CSS: Cascading Style Sheets
you browse to a long page on a slow connection and begin to scroll to read the content; while you are busy reading, the part of the page you are looking at suddenly jumps.
var() - CSS: Cascading Style Sheets
WebCSSvar
(doing so usually produces invalid syntax, or else a value whose meaning has no connection to the variable.) syntax the first argument to the function is the name of the custom property to be substituted.
Getting Started - Developer guides
} the full list of the readystate values is documented at xmlhttprequest.readystate and is as follows: 0 (uninitialized) or (request not initialized) 1 (loading) or (server connection established) 2 (loaded) or (request received) 3 (interactive) or (processing request) 4 (complete) or (request finished and response is ready) next, check the http response status codes of the http response.
Cross-browser audio basics - Developer guides
this is generally not a good option unless you can guarantee your users will have a fast network connection.
Mobile-friendliness - Developer guides
goal #3 (performance) “give your users a smooth experience, even on a slow connection.” though things have been getting better in recent years, browsing the internet over a wireless data connection can still be pretty painful.
<blockquote>: The Block Quotation element - HTML: Hypertext Markup Language
the connection topology is limited to a single point-to-point path for each carrier, used with standard carriers, but many carriers can be used without significant interference with each other, outside of early spring.
HTML documentation index - HTML: Hypertext Markup Language
WebHTMLIndex
229 link types: preconnect attribute, html, link, link types, reference the preconnect keyword for the rel attribute of the <link> element is a hint to browsers that the user is likely to need resources from the target resource's origin, and therefore the browser can likely improve the user experience by preemptively initiating a connection to that origin.
Link types: preconnect - HTML: Hypertext Markup Language
the preconnect keyword for the rel attribute of the <link> element is a hint to browsers that the user is likely to need resources from the target resource's origin, and therefore the browser can likely improve the user experience by preemptively initiating a connection to that origin.
Link types - HTML: Hypertext Markup Language
<link> <a>, <area>, <form> preconnect provides a hint to the browser suggesting that it open a connection to the linked web site in advance, without disclosing any private information or downloading any content, so that when the link is followed the linked content can be fetched more quickly.
HTTP authentication - HTTP
in the case of a "basic" authentication like shown in the figure, the exchange must happen over an https (tls) connection to be secure.
Identifying resources on the Web - HTTP
ecific file names ftp file transfer protocol http/https hyper text transfer protocol (secure) javascript url-embedded javascript code mailto electronic mail address ssh secure shell tel telephone urn uniform resource names view-source source code of the resource ws/wss (encrypted) websocket connections authority www.example.com is the domain name or authority that governs the namespace.
MIME types (IANA media types) - HTTP
d file: <input type="file" name="myfile" value="test.txt"></label> <button>send the file</button> </form> will send this message: post / http/1.1 host: localhost:8000 user-agent: mozilla/5.0 (macintosh; intel mac os x 10.9; rv:50.0) gecko/20100101 firefox/50.0 accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 accept-language: en-us,en;q=0.5 accept-encoding: gzip, deflate connection: keep-alive upgrade-insecure-requests: 1 content-type: multipart/form-data; boundary=---------------------------8721656041911415653955004498 content-length: 465 -----------------------------8721656041911415653955004498 content-disposition: form-data; name="mytextfield" test -----------------------------8721656041911415653955004498 content-disposition: form-data; name="mycheckbox" on ----------...
Reason: CORS request did not succeed - HTTP
the http request which makes use of cors failed because the http connection failed at either the network or protocol level.
Reason: CORS preflight channel did not succeed - HTTP
make sure your code only preflights once per connection.
Using HTTP cookies - HTTP
WebHTTPCookies
cookies are sent with every request, so they can worsen performance (especially for mobile data connections).
Access-Control-Allow-Headers - HTTP
nd origin, such as: options /resource/foo access-control-request-method: delete access-control-request-headers: origin, x-requested-with origin: https://foo.bar.org response if the server allows cors requests to use the delete method, it responds with an access-control-allow-methods response header, which lists delete along with the other methods it supports: http/1.1 200 ok content-length: 0 connection: keep-alive access-control-allow-origin: https://foo.bar.org access-control-allow-methods: post, get, options, delete access-control-max-age: 86400 if the requested method isn't supported, the server will respond with an error.
Alt-Svc - HTTP
WebHTTPHeadersAlt-Svc
if the cached entry expires, the client can no longer use this alternative service for new connections.
Cache-Control - HTTP
it therefore forbids a proxy or browser feature, such as google’s web light, from converting images to minimize data for a cache store or slow connection.
CSP: connect-src - HTTP
examples violation cases given this csp header: content-security-policy: connect-src https://example.com/ the following connections are blocked and won't load: <a ping="https://not-example.com"> <script> var xhr = new xmlhttprequest(); xhr.open('get', 'https://not-example.com/'); xhr.send(); var ws = new websocket("https://not-example.com/"); var es = new eventsource("https://not-example.com/"); navigator.sendbeacon("https://not-example.com/", { ...
Save-Data - HTTP
this could be for reasons such as high transfer costs, slow connection speeds, etc.
Sec-WebSocket-Accept - HTTP
that is, this is header is sent from server to client to inform that server is willing to initiate a websocket connection.
HTTP Public Key Pinning (HPKP) - HTTP
if an attacker is able to compromise a single ca, they can perform mitm attacks on various tls connections.
101 Switching Protocols - HTTP
WebHTTPStatus101
http/1.1 101 switching protocols upgrade: websocket connection: upgrade specifications specification title rfc 7231, section 6.2.2: 101 switching protocol hypertext transfer protocol (http/1.1): semantics and content ...
205 Reset Content - HTTP
WebHTTPStatus205
status 205 reset content specifications specification title rfc 7231, section 6.3.6: 205 reset content hypertext transfer protocol (http/1.1): semantics and content compatibility notes browser behavior differs if this response erroneously includes a body on persistent connections see 204 no content for more detail.
413 Payload Too Large - HTTP
WebHTTPStatus413
the http 413 payload too large response status code indicates that the request entity is larger than limits defined by server; the server might close the connection or return a retry-after header field.
426 Upgrade Required - HTTP
WebHTTPStatus426
status 426 upgrade required examples http/1.1 426 upgrade required upgrade: http/2.0 connection: upgrade content-length: 53 content-type: text/plain this service requires use of the http/2.0 protocol specifications specification title rfc 7231, section 6.5.15: 426 upgrade required hypertext transfer protocol (http/1.1): semantics and content ...
Media container formats (file types) - Web media technologies
consider using this container format general purpose video, preferably in an open format webm (ideally with mp4 fallback) general purpose video mp4 (ideally with webm or ogg fallback) high compression optimized for slow connections 3gp (ideally with mp4 fallback) compatibility with older devices/browsers quicktime (ideally with avi and/or mpeg-2 fallback) these suggestions make a number of assumptions.
Image file type and format guide - Web media technologies
this is particularly useful when network connections are slow.
Handling media support issues in web content - Web media technologies
a similar concept can be applied to still images; if an image you wish to present is very large and may take time to download (especially for slower devices or connections), you can offer a lower-resolution or alternate version that will be displayed until the full-quality version is available to be displayed.
Progressive web app structure - Progressive web apps (PWAs)
it also allows the website to be accessible offline if the network connection is not available.
The building blocks of responsive design - Progressive web apps (PWAs)
mobiles in general (more commonly in some parts of the world than others) are on lower bandwidth connections and have less memory available than desktop devices, so yes, those extra kilobytes really do count.
Structural overview of progressive web apps - Progressive web apps (PWAs)
it also allows the website to be accessible offline if the network connection is not available.
<metadata> - SVG: Scalable Vector Graphics
WebSVGElementmetadata
example <svg width="400" viewbox="0 0 400 300" xmlns="http://www.w3.org/2000/svg" xmlns:xlink="http://www.w3.org/1999/xlink"> <metadata> <rdf:rdf xmlns:rdf="http://www.w3.org/1999/02/22-rdf-syntax-ns#" xmlns:connect="http://www.w3.org/1999/08/29-svg-connections-in-rdf#"> <rdf:description about="#cablea"> <connect:ends rdf:resource="#socket1"/> <connect:ends rdf:resource="#computera"/> </rdf:description> <rdf:description about="#cableb"> <connect:ends rdf:resource="#socket2"/> <connect:ends rdf:resource="#computerb"/> </rdf:description> <rdf:description about="#cablen"> <connect:ends ...
Mixed content - Web security
when a user visits a page served over https, their connection with the web server is encrypted with tls and is therefore safeguarded from most sniffers and man-in-the-middle attacks.
Same-origin policy - Web security
even if you use only secure https connections, any cookie you see may have been set using an insecure connection.
Tutorials
tips for authoring fast-loading html pages optimize web pages to provide a more responsive site for visitors and reduce the load on your web server and internet connection.