RTCIceTransport: statechange event - Web APIs
a statechange event occurs when the rtcicetransport changes state. the state can be used to determine how far through the process of examining, verifying, and selecting a valid candidate pair is prior to successfully connecting the two peers for webrtc communications. 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. ...
...o successfully connecting the two peers for webrtc communications. 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. ...
...wo peers for webrtc communications. 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. ...
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Introduction to the Real-time Transport Protocol (RTP) - Web APIs
keeping latency to a minimum is especially important for webrtc, since face-to-face communication needs to be performed with as little latency as possible. the more time lag there is between one user saying something and another hearing it, the more likely there is to be episodes of cross-talking and other forms of confusion. ...
... keeping latency to a minimum is especially important for webrtc, since face-to-face communication needs to be performed with as little latency as possible. the more time lag there is between one user saying something and another hearing it, the more likely there is to be episodes of cross-talking and other forms of confusion. ...
... communication needs to be performed with as little latency as possible. the more time lag there is between one user saying something and another hearing it, the more likely there is to be episodes of cross-talking and other forms of confusion. key features of rtp before examining rtp's use in webrtc contexts, it's useful to have a general idea of what rtp does and does not offer. ...
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DTLS (Datagram Transport Layer Security) - MDN Web Docs Glossary: Definitions of Web-related terms
datagram transport layer security (dtls) is a protocol used to secure datagram-based communications. it's based on the stream-focused transport layer security (tls), providing a similar level of security. as a datagram protocol, dtls doesn't guarantee the order of message delivery, or even that messages will be delivered at all. however, dtls gains the benefits of datagram protocols, too; in particular, the lower overhead and reduced latency. these features are especially useful for one of the most common areas in which dtls comes into play: webrtc. all of the webrtc related protocols are required to encrypt their communications using dtls; this includes sctp, srtp, and stun. ...
.... however, dtls gains the benefits of datagram protocols, too; in particular, the lower overhead and reduced latency. these features are especially useful for one of the most common areas in which dtls comes into play: webrtc. all of the webrtc related protocols are required to encrypt their communications using dtls; this includes sctp, srtp, and stun. ...
... the lower overhead and reduced latency. these features are especially useful for one of the most common areas in which dtls comes into play: webrtc. all of the webrtc related protocols are required to encrypt their communications using dtls; this includes sctp, srtp, and stun. ...
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nsITransport
netwerk/base/public/nsitransport.idlscriptable this interface provides a common way of accessing i/o streams connected to some resource. this interface does not in any way specify the resource. inherits from: nsisupports last changed in gecko 1.8 (firefox 1.5 / thunderbird 1.5 / seamonkey 1.0) this interface provides methods to open blocking or non-blocking, buffered or unbuffered streams to the resource. the name "transport" is meant to connote the inherent data transfer implied by this interface (that is, data is being transfered in some fashion via the streams exposed by this interface). a transport can have an event sink associated with it. the event sink receives transport-specific events as the transfer is occurring. ...
... non-blocking, buffered or unbuffered streams to the resource. the name "transport" is meant to connote the inherent data transfer implied by this interface (that is, data is being transfered in some fashion via the streams exposed by this interface). a transport can have an event sink associated with it. the event sink receives transport-specific events as the transfer is occurring. for a socket transport, these events can include status about the connection. see nsisockettransport for more info about socket transport specifics. method overview void close(in nsresult areason); nsiinputstream openinputstream(in unsigned long aflags, in unsigned long asegmentsize, in unsigned long asegmentcount); nsioutputstream openoutputstream(in unsigned long aflags, in unsigned long asegmentsize, in unsig...
...ion. see nsisockettransport for more info about socket transport specifics. method overview void close(in nsresult areason); nsiinputstream openinputstream(in unsigned long aflags, in unsigned long asegmentsize, in unsigned long asegmentcount); nsioutputstream openoutputstream(in unsigned long aflags, in unsigned long asegmentsize, in unsigned long asegmentcount); void seteventsink(in nsitransporteventsink asink, in nsieventtarget aeventtarget); constants open flags. constant value description open_blocking 1<<0 open flags. open_unbuffered 1<<1 generic nsitransporteventsink status codes. nsitransport implementations may override these status codes with their own more specific status codes (for example, see nsisockettransport). constant value description status_reading 0...
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AuthenticatorAttestationResponse.getTransports() - Web APIs
gettransports() is a method of the authenticatorattestationresponse interface that returns an array containing strings describing the different transports which may be used by the authenticator. such transports may be usb, nfc, ble or internal (applicable when the authenticator is not removable from the device). note: an authenticatorattestationresponse instance is available on publickeycredential.response after calling navigator.credentials.create(). note: this method may only be used in top-level contexts and will not be available in an <iframe> for example. syntaxe arrtransports = authenticatorattestationresponse.gettransports() parameters none. ...
...note: an authenticatorattestationresponse instance is available on publickeycredential.response after calling navigator.credentials.create(). note: this method may only be used in top-level contexts and will not be available in an <iframe> for example. syntaxe arrtransports = authenticatorattestationresponse.gettransports() parameters none. return value an array containing the different transports supported by the authenticator or nothing if this information is not available.of the processing of the different extensions by the client. the elements of this array are supposed to be in lexicographical order. their values may be : "usb": the authenticator can be contacted via a removable usb link "nfc": the authenticator may be used over nfc (near field communication) "ble": the authenti...
... rec...
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RTCIceTransport.component - Web APIs
this value of rtcicecomponent corresponds to the component id 2. ...
...this value of rtcicecomponent corresponds to the component id 2. ...
...this value of rtcicecomponent corresponds to the component id 2. ...
...And 3 more matches

RTCIceTransport.getRemoteCandidates() - Web APIs
return value an array containing one rtcicecandidate object for each candidate that has been received so far from the remote peer during the current ice candidate gathering session. it's important to keep in mind that there's no way to correlate these remote candidates with compatible local candidates. ...
...idates() returns an array which contains one rtcicecandidate for each of the candidates that have been received from the remote peer so far during the current ice gathering session. 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. syntax remotecandidates = rtcicetransport.getremotecandidates(); parameters none. return value an array containing one rtcicecandidate object for each candidate that has been received so far from the remote peer during the current ice candidate gathering session. it's important to keep in mind that there's no way to correlate these remote candidates with compatible local candidates. to find the best match found so far, call rtcicetr...
...h candidate that has been received so far from the remote peer during the current ice candidate gathering session. it's important to keep in mind that there's no way to correlate these remote candidates with compatible local candidates. to find the best match found so far, call rtcicetransport.getselectedcandidatepair(). ...
... correlate these remote candidates with compatible local candidates. to find the best match found so far, call rtcicetransport.getselectedcandidatepair(). 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. var remotecandidates = pc.getsenders()[0].transport.transport.getremotecandidates(); remotecandidates.foreach(function(candidate, index)) { console.log("candidate " + index + ": " + candidate.candidate); }); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecandidate.getremotecandidates()' in that specification. candidate recommendation initia...

RTCIceTransport.ongatheringstatechange - Web APIs
the ongatheringstatechange event handler for the rtcicetransport interface specifies an event handler that is to be called when the gatheringstatechange event occurs on the transport. this event is delivered whenever the transport's gatheringstate property changes. syntax rtcicetransport.ongatheringstatechange = statechangehandler; value a function to be called when the rtcicetransport object's gathering state changes. to determine the new state, examine the value of gatheringstate. the gathering state indicates whether or not the ice agent has begun gathering candidates, and if so, whether or not gathering has finished. ...
...etransport.ongatheringstatechange = statechangehandler; value a function to be called when the rtcicetransport object's gathering state changes. to determine the new state, examine the value of gatheringstate. the gathering state indicates whether or not the ice agent has begun gathering candidates, and if so, whether or not gathering has finished. its possible values are: "new" the rtcicetransport is newly created and has not yet started to gather ice candidates. "gathering" the transport is in the process of gathering candidates. "complete" the transport has finished gathering ice candidates and has sent the end-of-candidates indicator to the remote device. ...
...athering candidates, and if so, whether or not gathering has finished. its possible values are: "new" the rtcicetransport is newly created and has not yet started to gather ice candidates. "gathering" the transport is in the process of gathering candidates. "complete" the transport has finished gathering ice candidates and has sent the end-of-candidates indicator to the remote device. the transport won't gather any further candidates unless an ice restart occurs, at which point the gathering process starts over from scratch. ...
...e. the transport won't gather any further candidates unless an ice restart occurs, at which point the gathering process starts over from scratch. example this snippet establishes a handler for the gatheringstatechange event that checks to see if the state has changed to "complete", indicating that all ice candidates from both the local and remote peers have been received and processed. var icetransport = pc.getsenders()[0].transport.transport; icetransport.ongatheringstatechange = function(event) { if (icetransport.gatheringstate == "complete") { allcandidatesreceived(pc); } } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicetransport.ongatheringstatechange' in that specification. ...

nsITransportEventSink
netwerk/base/public/nsitransport.idlscriptable implemented by clients that wish to receive transport events. inherits from: nsisupports last changed in gecko 1.7 method overview void ontransportstatus(in nsitransport atransport, in nsresult astatus, in unsigned long long aprogress, in unsigned long long aprogressmax); methods ontransportstatus() transport status notification. void ontransportstatus( in nsitransport atransport, in nsresult astatus, in unsigned long long aprogress, in unsigned long long aprogressmax ); parameters atransport the transport sending this status notification. astatus the transport status (resolvable to a string using nsierrorservice). see nsisockettransport for socket specific status codes and more comments. aprogress the amount of data eit...
...e transport status (resolvable to a string using nsierrorservice). see nsisockettransport for socket specific status codes and more comments. aprogress the amount of data either read or written depending on the value of the status code. this value is relative to aprogressmax. aprogressmax the maximum amount of data that will be read or written. if unknown, 0xffffffff will be passed. ...
...(resolvable to a string using nsierrorservice). see nsisockettransport for socket specific status codes and more comments. aprogress the amount of data either read or written depending on the value of the status code. this value is relative to aprogressmax. aprogressmax the maximum amount of data that will be read or written. if unknown, 0xffffffff will be passed. ...

RTCIceTransport.getLocalCandidates() - Web APIs
the rtcicetransport method getlocalcandidates() returns an array of rtcicecandidate objects, one for each of the candidates that have been gathered by the local device during the current ice agent session. the local candidates are placed in this list by the ice agent prior to being delivered to the local client's code in an icecandidate event so that the client can forward the candidates to the remote peer. syntax localcandidates = rtcicetransport.getlocalcandidates(); parameters none. return value a javascript array containing one rtcicecandidate object for each candidate that has been identified so far during the ice candidate gathering session. you can't correlate these local candidates with matching remote candidates. ...
...tes to the remote peer. syntax localcandidates = rtcicetransport.getlocalcandidates(); parameters none. return value a javascript array containing one rtcicecandidate object for each candidate that has been identified so far during the ice candidate gathering session. you can't correlate these local candidates with matching remote candidates. to find the best match found so far, call rtcicetransport.getselectedcandidatepair(). 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. ...
...an't correlate these local candidates with matching remote candidates. to find the best match found so far, call rtcicetransport.getselectedcandidatepair(). 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. var localcandidates = pc.getsenders()[0].transport.transport.getlocalcandidates(); localcandidates.foreach(function(candidate, index)) { console.log("candidate " + index + ": " + candidate.candidate); }); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecandidate.getlocalcandidates()' in that specification. candidate recommendation initial...

RTCIceTransport.onstatechange - Web APIs
the onstatechange event handler for the rtcicetransport interface is a property which specifies a function to serve as the eventhandler for the statechange event that is fired whenever the transport's state changes. syntax rtcicetransport.onstatechange = statechangehandler; value set this property to reference a function you provide that is called by the webrtc layer when the rtcicetransport object's state changes. the event handler receives as its sole input an event object describing the statechange event which occurred. to determine the new state, examine the value of state. ...
...et this property to reference a function you provide that is called by the webrtc layer when the rtcicetransport object's state changes. the event handler receives as its sole input an event object describing the statechange event which occurred. to determine the new state, examine the value of state. 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. ...
...nt handler receives as its sole input an event object describing the statechange event which occurred. to determine the new state, examine the value of state. 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. var icetransport = pc.getsenders()[0].transport.icetransport; icetransport.onstatechange = function(event) { if (icetransport.state == "failed") { handlefailure(pc); } } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicetransport.onstatechange' in that specification. candidate recommendation initia...

Transport Layer Security - Web security
applications that use tls can choose their security parameters, which can have a substantial impact on the security and reliability of data. ...
...applications that use tls can choose their security parameters, which can have a substantial impact on the security and reliability of data. ...
... technology needed to be specified to ensure interoperability among all web browsers and servers. the internet engineering task force (ietf) specified tls 1.0 in rfc 2246 in january, 1999. the current version of tls is 1.3 (rfc 8446). despite the fact that the web now uses tls for encryption, many people still refer to it as "ssl" out of habit. although tls can be used on top of any low-level transport protocol, the original goal of the protocol was to encrypt http traffic. http encrypted using tls is commonly referred to as https. tls-encrypted web traffic is by convention exchanged on port 443 by default, while unencrypted http uses port 80 by default. https remains an important use case for tls. http over tls tls provides three primary services that help ensure the safety and security of d...

RTP (Real-time Transport Protocol) and SRTP (Secure RTP) - MDN Web Docs Glossary: Definitions of Web-related terms
the real-time transport protocol (rtp) is a network protocol which described how to transmit various media (audio, video) from one endpoint to another in a real-time fashion. rtp is suitable for video-streaming application, telephony over ip like skype and conference technologies. the secure version of rtp, srtp, is used by webrtc, and uses encryption and authentication to minimize the risk of denial-of-service attacks and security breaches. rtp is rarely used alone; instead, it is used in conjunction with other protocols like rtsp and sdp. ...
...plication, telephony over ip like skype and conference technologies. ...

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. read only. securitystate unsigned long a flag detailing the security state of the connection. the possible values are defined in nsiwebprogresslistener. read only. shortsecuritydescription wstring for secure connections (ssl) gives the common name (cn) of the certifying authority. read only. ...
...te type description errormessage wstring error message on connection failure. read only. securitystate unsigned long a flag detailing the security state of the connection. the possible values are defined in nsiwebprogresslistener. read only. shortsecuritydescription wstring for secure connections (ssl) gives the common name (cn) of the certifying authority. read only. see also nsisockettransport ...

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. its value must come from the rtcicetransportpolicy enumerated type. if this property isn't included in the rtcconfiguration, the default value, all, is used. syntax let rtcconfiguration = { icetransportpolicy: policy }; rtcconfiguration.icetransportpolicy = policy; let policy = rtcconfiguration.icetransportpolicy; value a domstring which indicates what ice candidate policy the ice agent should use during the negotiation process, per the jsep standard. the permitted values are: all the ice agent is permitted to use any kind of candidate, including both local and relay candidates. the agent—or ...
...nt in some security situations. for example, in a video calling application, the app may want to prevent unknown callers from learning the callee's ip addresses until the callee has agreed to receive the call. 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. candidate recommendation initial definition. ...

RTCIceTransport.gatheringState - Web APIs
the read-only rtcicetransport property gatheringstate returns a domstring from the enumerated type rtcicegathererstate that indicates what gathering state the ice agent is currently in: "new", "gathering", or "complete". syntax gatherstate = rtcicetransport.gatheringstate; value a string from the rtcicegathererstate enumerated type whose value indicates the current state of the ice agent's candidate gathering process: "new" the rtcicetransport is newly created and has not yet started to gather ice candidates. "gathering" the transport is in the process of gathering candidates. "complete" the transport has finished gathering ice candidates and has sent the end-of-candidates indicator to the remote device. ...
...ed type whose value indicates the current state of the ice agent's candidate gathering process: "new" the rtcicetransport is newly created and has not yet started to gather ice candidates. "gathering" the transport is in the process of gathering candidates. "complete" the transport has finished gathering ice candidates and has sent the end-of-candidates indicator to the remote device. the transport won't gather any further candidates unless an ice restart occurs, at which point the gathering process starts over from scratch. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicetransport.gatheringstate' in that specification. candidate recommendation initial definition. ...

RTCIceTransport.getLocalParameters() - Web APIs
the rtcicetransport method getlocalparameters() returns an rtciceparameters object which provides information uniquely identifying the local peer for the duration of the ice session. the local peer's parameters are obtained during ice signaling and delivered to the transport when the client calls rtcpeerconnection.setlocaldescription(). syntax parameters = rtcicetransport.getlocalparameters(); parameters none. return value an rtciceparameters object indicating the usernamefragment and password which uniquely identify the local peer for the duration of the ice session. returns null if the parameters have not yet been received. ...
...ers(); parameters none. return value an rtciceparameters object indicating the usernamefragment and password which uniquely identify the local peer for the duration of the ice session. returns null if the parameters have not yet been received. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicetransport.getlocalparameters' in that specification. candidate recommendation initial specification. ...

RTCIceTransport.getRemoteParameters() - Web APIs
the rtcicetransport method getremoteparameters() returns an rtciceparameters object which provides information uniquely identifying the remote peer for the duration of the ice session. the remote peer's parameters are received during ice signaling and delivered to the transport when the client calls rtcpeerconnection.setremotedescription(). syntax parameters = rtcicetransport.getremoteparameters(); parameters none. return value an rtciceparameters object indicating the usernamefragment and password which uniquely identify the remote peer for the duration of the ice session. returns null if the parameters haven't been received yet. ...
...ers(); parameters none. return value an rtciceparameters object indicating the usernamefragment and password which uniquely identify the remote peer for the duration of the ice session. returns null if the parameters haven't been received yet. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicetransport.getremoteparameters' in that specification. candidate recommendation initial specification. ...

RTCIceTransport.role - Web APIs
the read-only rtcicetransport property role indicates which ice role the transport is fulfilling: that of the controlling agent, or the agent that is being controlled. you can learn more about ice roles in choosing a candidate pair in webrtc connectivity. syntax icerole = rtcicetransport.role; value a domstring specifying whether the rtcicetransport represents the controlling agent or the controlled agent. the value must be one of those found in the enumerated type rtcicerole: "controlling" the rtcicetransport object is serving as the controlling agent. "controlled" the transport is the controlled agent. ...
... controlling agent or the controlled agent. the value must be one of those found in the enumerated type rtcicerole: "controlling" the rtcicetransport object is serving as the controlling agent. "controlled" the transport is the controlled agent. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicetransport.role' in that specification. candidate recommendation initial definition. ...

RTCRtpReceiver.transport - Web APIs
the read-only transport property of an rtcrtpreceiver object provides the rtcdtlstransport object used to interact with the underlying transport over which the receiver is exchanging real-time transport control protocol (rtcp) packets. this transport is responsible for receiving the data for the media on the receiver's track. syntax let transport = rtcrtpreceiver.transport; value an rtcdtlstransport object representing the underlying transport being used by the receiver to exchange packets with the remote peer, or null if the receiver isn't yet connected to a transport. description when the rtcrtpreceiver is first created, the value of transport is null. this is replaced with an rtcdtlstransport once the receiver's transport has been established. note that when bundling is in effect—that...
...port. description when the rtcrtpreceiver is first created, the value of transport is null. this is replaced with an rtcdtlstransport once the receiver's transport has been established. 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. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpreceiver.transport' in that specification. candidate recommendation ...

RTCRtpSender.transport - Web APIs
the read-only transport property of an rtcrtpsender object provides the rtcdtlstransport object used to interact with the underlying transport over which the sender is exchanging real-time transport control protocol (rtcp) packets. this transport is responsible for receiving the data for the media on the sender's track. syntax let transport = rtcrtpsender.transport; value an rtcdtlstransport object representing the underlying transport being used by the sender to exchange packets with the remote peer, or null if the sender isn't yet connected to a transport. description when the rtcrtpsender is first created, the value of transport is null. this is replaced with an rtcdtlstransport once the sender's transport has been established. note that when bundling is in effect—that is, when the rt...
... transport. description when the rtcrtpsender is first created, the value of transport is null. this is replaced with an rtcdtlstransport once the sender's transport has been established. 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. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpsender.transport' in that specification. candidate recommendation ...

Transport Layer Security (TLS) - MDN Web Docs Glossary: Definitions of Web-related terms
both ssl and tls are client / server protocols that ensure communication privacy by using cryptographic protocols to provide security over a network. when a server and client communicate using tls, it ensures that no third party can eavesdrop or tamper with any message. 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. ...

RTCIceCandidateStats.mozLocalTransport - Web APIs
the non-standard mozilla extension to the rtcicecandidatestats dictionary, mozlocaltransport, has been supplanted by the standard relayprotocol property. syntax instead of using mozlocaltransport, you should use code like this: localtransport = rtcicecandidatestats.relayprotocol; specifications not part of any specification. ...

RTCIceCandidateStats.transportId - Web APIs
the rtcicecandidatestats dictionary's transportid property is a string that uniquely identifies the transport that produced the rtctransportstats from which information about this candidate was taken. syntax transportid = rtcicecandidatestats.transportid; value a domstring whose value uniquely identifies the transport from which any transport-related information accumulated in the rtcicecandidatestats was taken. this can be used to compare candidates that would use the same transport, for example. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcicecandidatestats.transportid' in that specification. candidate recommendation initial specification. ...

RTCRtpStreamStats.transportId - Web APIs
the rtcrtpstreamstats dictionary's transportid property is a string which uniquely identifies the object from which the statistics contained in the rtctransportstats properties in the rtcstatsreport. syntax var transportid = rtcrtpstreamstats.transportid; value a domstring uniquely identifying the source of the statistics contained the rtctransportstats properties in the rtcstatsreport. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcrtpstreamstats.transportid' in that specification. candidate recommendation initial definition. ...

RTCSctpTransport.state - Web APIs
the state read-only property of the rtcsctptransport interface provides information which describes a stream control transmission protocol (sctp) transport state. syntax var mystate = sctptransport.state; value a string whose value is taken from the rtcsctptransportstate enumerated type. 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. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcsctptransport.state' in that specification. candidate recommendation initial definition. ...

Index - Web APIs
testationresponse, property, reference, web authentication api, webauthn the attestationobject property of the authenticatorattestationresponse interface returns an arraybuffer containing the new public key, as well as signature over the entire attestationobject with a private key that is stored in the authenticator when it is manufactured. 222 authenticatorattestationresponse.gettransports() api, authenticatorattestationresponse, method, reference, web authentication api, webauthn gettransports() is a method of the authenticatorattestationresponse interface that returns an array containing strings describing the different transports which may be used by the authenticator. 223 authenticatorresponse api, authentication, authenticatorresponse, interface, refe...
...cific resource, such as an xmlhttprequest, <svg>, image, or script. 3030 performanceresourcetiming.connectend api, property, reference, web performance 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. 3031 performanceresourcetiming.connectstart api, property, reference, web performance the connectstart read-only property returns the timestamp immediately before the user agent starts establishing the connection to the server to retrieve the resource. 3032 performanceresour...
...ad-only property returns a timestamp representing the start time of the fetch which that initiates the redirect. 3041 performanceresourcetiming.requeststart api, property, reference, web performance the requeststart read-only property returns a timestamp of the time immediately before the browser starts requesting the resource from the server, cache, or local resource. ...
...And 53 more matches

RTCPeerConnection - Web APIs
descriptions in webrtc connectivity.remotedescription read only 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. if this hasn't been set yet, this is null.sctp 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. if sctp hasn't been negotiated, this value is null.signalingstate read only 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...
...restartice() method to allow a web application to easily request that ice candidate gathering be redone on both ends of the connection.setconfiguration() 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.setidentityprovider() 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. the idp will be used only when an assertion is needed.setlocaldescription()the rtcpeerconnection method setlocaldescription() changes the local descripti...
...k events on each stream within the rtcpeerconnection. also available as the onaddstream event handler property. constants rtcbundlepolicy enum the rtcbundlepolicy enum defines string constants which are used to request a specific policy for gathering ice candidates if the remote peer isn't "bundle-aware" (compatible with the sdp bundle standard for bundling multiple media streams on a single transport link). all browser implementations are bundle-aware. if the remote endpoint is bundle-aware, all media tracks and data channels are bundled onto a single transport at the completion of negotiation, regardless of policy used, and any superfluous transports that were created initially are closed at that point. note: in technical terms, a bundle lets all media flow between two peers flow across a...
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TCP/IP Security - Archive of obsolete content
he layers to its destination. essentially, the data produced by a layer is encapsulated in a larger container by the layer below it. the four tcp/ip layers, from highest to lowest, are shown below. application layer. this layer sends and receives data for particular applications, such as domain name system (dns), hypertext transfer protocol (http), and simple mail transfer protocol (smtp). transport layer. this layer provides connection-oriented or connectionless services for transporting application layer services between networks. the transport layer can optionally assure the reliability of communications. transmission control protocol (tcp) and user datagram protocol (udp) are commonly used transport layer protocols. network layer. this layer routes packets across networks. internet...
...ion data, they cannot protect tcp/ip information such as ip addresses because this information exists at a lower layer. whenever possible, application layer controls for protecting network communications should be standards-based solutions that have been in use for some time. one example is secure multipurpose internet mail extensions (s/mime), which is commonly used to encrypt email messages. transport layer. controls at this layer can be used to protect the data in a single communication session between two hosts. because ip information is added at the network layer, transport layer controls cannot protect it. the most common use for transport layer protocols is securing http traffic; the transport layer security (tls) protocol is usually used for this. (tls is the standards-based version ...
...nsport layer protocols is securing http traffic; the transport layer security (tls) protocol is usually used for this. (tls is the standards-based version of ssl version 3. more information on tls is available in rfc 4346, the tls protocol version 1.1, available at https://www.ietf.org/rfc/rfc4346.txt. ...
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Index - MDN Web Docs Glossary: Definitions of Web-related terms
102 dom (document object model) codingscripting, dom, glossary the dom (document object model) is an api that represents and interacts with any html or xml document. the dom is a document model loaded in the browser and representing the document as a node tree, where each node represents part of the document (e.g. an element, text string, or comment). 103 dtls (datagram transport layer security) dtls, intro datagram transport layer security (dtls) is a protocol used to secure datagram-based communications. it's based on the stream-focused transport layer security (tls), providing a similar level of security. as a datagram protocol, dtls doesn't guarantee the order of message delivery, or even that messages will be delivered at all. however, dtls gains the benefi...
...y hmac is a protocol used for cryptographically authenticating messages. 192 hpkp glossary, security http public key pinning (hpkp) is a security feature that tells a web client to associate a specific cryptographic public key with a certain web server to decrease the risk of mitm attacks with forged certificates. 193 hsts http, security http strict transport security lets a web site inform the browser that it should never load the site using http and should automatically convert all attempts to access the site using http to https requests instead. it consists in one http header, strict-transport-security, sent by the server with the resource. 194 html codingscripting, glossary, html, l10n:priority html (hypertext markup language)...
...ingscripting, glossary, language, python, programming python is a high level general-purpose programming language. it uses a multi-paradigm approach, meaning it supports procedural, object-oriented, and some functional programming constructs. 348 quic glossary, http, quic, reference, web performance quick udp internet connection, or quic, is an experimental multiplexed transport protocol implemented on udp. it was developed by google as a way to experiment with ways to improve tcp and web application delivery. 349 quality values glossary, webmechanics quality values, or q-values and q-factors, are used to describe the order of priority of values in a comma-separated list. it is a special syntax allowed in some http headers and in html. 350 ...
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WebRTC API - Web APIs
atsreport provides information detailing statistics for a connection or for an individual track on the connection; the report can be obtained by calling rtcpeerconnection.getstats(). details about using webrtc statistics can be found in webrtc statistics api. rtcicecandidate represents a candidate interactive connectivity establishment (ice) server for establishing an rtcpeerconnection. rtcicetransport represents information about an ice transport. rtcpeerconnectioniceevent represents events that occur in relation to ice candidates with the target, usually an rtcpeerconnection. only one event is of this type: icecandidate. rtcrtpsender manages the encoding and transmission of data for a mediastreamtrack on an rtcpeerconnection. rtcrtpreceiver manages the reception and decoding of data fo...
...amtrack on an rtcpeerconnection. rtcrtpreceiver manages the reception and decoding of data for a mediastreamtrack on an rtcpeerconnection. rtctrackevent the interface used to represent a track event, which indicates that an rtcrtpreceiver object was added to the rtcpeerconnection object, indicating that a new incoming mediastreamtrack was created and added to the rtcpeerconnection. rtcsctptransport provides information which describes a stream control transmission protocol (sctp) transport and 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. dictionaries rtcconfiguration used to provide configuration options for an rtcpeerconnection. rtciceser...
...ource contributed was played out. events bufferedamountlow the amount of data currently buffered by the data channel—as indicated by its bufferedamount property—has decreased to be at or below the channel's minimum buffered data size, as specified by bufferedamountlowthreshold. close the data channel has completed the closing process and is now in the closed state. its underlying data transport is completely closed at this point. you can be notified before closing completes by watching for the closing event instead. closing the rtcdatachannel has transitioned to the closing state, indicating that it will be closed soon. you can detect the completion of the closing process by watching for the close event. connectionstatechange the connection's state, which can be accessed in connecti...
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NSPR Poll Method
of prfiledesc. the poll method is not to be confused with the pr_poll function. the poll method operates on a single netscape portable runtime (nspr) file descriptor, whereas pr_poll operates on a collection of nspr file descriptors. pr_poll uses the poll method behind the scene, but it is also possible to use the poll method directly. we consider a stack of nspr i/o layers on top of the network transport. each i/o layer is represented by a prfiledesc structure and the protocol of that layer is implemented by a priomethods table. the bottom layer is a wrapper for the underlying network transport. the nspr library provides a reference implementation of the bottom layer using the sockets api, but you can provide your own implementation of the bottom layer using another network transport api. the pol...
...the arguments and return value of the poll method are described below. ...
...vents at the top layer of the i/o layer stack that the caller is interested in. for pr_recv, you should pass pr_poll_read as the in_flags argument to the poll method for pr_send, you should pass pr_poll_write as the in_flags argument to the poll method out_flags [output argument] if an i/o layer is ready to satisfy the i/o request defined by in_flags without involving the underlying network transport, its poll method sets the corresponding event in *out_flags on return. for example, consider an i/o layer that buffers input data. if the caller wishes to test for read ready (that is, pr_poll_read is set in in_flags) and the layer has input data buffered, the poll method would set the pr_poll_read event in *out_flags. it can determine that without asking the underlying network transport. the cur...
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HTTP logging
all programs" submenu. on all newer versions of windows, you can hold down the windows key and press "r". type cmd and press enter, a new command prompt window with a black background will appear. copy and paste the following lines one at a time into the command prompt window. ...
...y and paste the following lines one at a time into the command prompt window. press the enter key after each one.: for 64-bit windows: set moz_log=timestamp,rotate:200,nshttp:5,cache2:5,nssockettransport:5,nshostresolver:5,cookie:5 set moz_log_file=%temp%\log.txt "c:\program files\mozilla firefox\firefox.exe" for 32-bit windows: set moz_log=timestamp,rotate:200,nshttp:5,cache2:5,nssockettransport:5,nshostresolver:5,cookie:5 set moz_log_file=%temp%\log.txt "c:\program files (x86)\mozilla firefox\firefox.exe" (these instructions assume that you installed firefox to the default location, and that drive c: is your windows startup disk. make the appropriate adjustments if those aren't the case.) reproduce whatever problem it is that you're having. once you've reproduced the problem, exit f...
...re quickly. linux this section offers information on how to capture http logs for firefox running on linux. quit out of firefox if it's running. open a new shell. the commands listed here assume a bash-compatible shell. copy and paste the following commands into the shell one at a time. make sure to hit enter after each line. export moz_log=timestamp,rotate:200,nshttp:5,cache2:5,nssockettransport:5,nshostresolver:5,cookie:5 export moz_log_file=/tmp/log.txt cd /path/to/firefox ./firefox reproduce the problem you're debugging. when the problem has been reproduced, exit firefox and look for the generated log files, which you can find at /tmp/log.txt. mac os x these instructions show how to log http traffic in firefox on mac os x. quit firefox is if it's currently running, by using...
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Index
t/hapticfeedback;1 as a service: 679 nsihttpactivitydistributor http, interfaces, interfaces:scriptable, xpcom, xpcom interface reference implemented by: mozilla.org/network/http-activity-distributor;1 as a service: 680 nsihttpactivityobserver http, interfaces, interfaces:scriptable, xpcom, xpcom interface reference called when activity occurs on the http transport. you should implement this method to perform whatever tasks you wish to perform when http activity occurs. 681 nsihttpchannel channels, http, interface reference, interfaces, interfaces:scriptable, network, xpcom, xpcom api reference, xpcom interface reference to create an http channel, use nsiioservice with a http uri, for example: 682 nsihttpchannelinternal h...
...référence(2), sockets, xpcom interface reference implemented by: @mozilla.org/network/socket;2?type=foo. to create an instance, use: 915 nsisocketproviderservice interfaces, interfaces:scriptable, sockets, xpcom interface reference given a string representing a socket type, this method returns an nsisocketprovider representing that socket type. 916 nsisockettransport interfaces, interfaces:scriptable, xpcom, xpcom api reference, xpcom interface reference to create an instance, call nsisockettransportservice.createtransport() 917 nsisockettransportservice interfaces, interfaces:scriptable, xpcom, xpcom api reference, xpcom interface reference implemented by: @mozilla.org/network/socket-transport-service;1. to create an instance, ...
...ion manager's notification list. listeners are notified whenever a transaction is done, undone, or redone. the listener's nsitransactionlistener.addref() method is called. 1023 nsitransferable gecko, interface, mozilla, reference, référence(2), xpcom, xpcom interface reference implemented by: @mozilla.org/widget/transferable;1. ...
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RTCConfiguration.bundlePolicy - Web APIs
tion() 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. this string, which must be a member of the rtcbundlepolicy enumeration, has the following possible values: balanced the ice agent begins by creating one rtcdtlstransport to handle each type of content added: one for audio, one for video, and one for the rtc data channel, if applicable. if the remote peer isn't bundle-aware, the ice agent chooses one audio track and one video track and those two tracks are each assigned to the corresponding rtcdtlstransport. all other tracks are ignored by the connection. this is the default, and most compatible, policy. max-comp...
... audio, one for video, and one for the rtc data channel, if applicable. if the remote peer isn't bundle-aware, the ice agent chooses one audio track and one video track and those two tracks are each assigned to the corresponding rtcdtlstransport. all other tracks are ignored by the connection. this is the default, and most compatible, policy. max-compat the ice agent intially creates one rtcdtlstransport for each media track and a separate one for the rtcdatachannel, if one is created. ...
...if the remote peer isn't bundle-compatible, only one media track is negotiated and the rest are ignored. ...
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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. the options include ice server and transport settings and identity information. properties bundlepolicy optional specifies how to handle negotiation of candidates when the remote peer is not compatible with the sdp bundle standard. this must be one of the values from the enum rtcbundlepolicy. if this value isn't included in the dictionary, "balanced" is assumed. certificates optional an array of objects of type rtccertificate which a...
...nging the size of the ice candidate pool may trigger the beginning of ice gathering. iceservers optional an array of rtciceserver objects, each describing one server which may be used by the ice agent; these are typically stun and/or turn servers. if this isn't specified, the connection attempt will be made with no stun or turn server available, which limits the connection to local peers. icetransportpolicy optional the current ice transport policy; this must be one of the values from the rtcicetransportpolicy enumeration. if the policy isn't specified, all is assumed by default, allowing all candidates to be considered. a value of relay limits the candidates to those relayed through another server, such as a stun or turn server. peeridentity optional a domstring which specifies the target ...
...t non-multiplexed rtcp. the value must be one of those from the rtcrtcpmuxpolicy enum. the default is "require". constants rtcbundlepolicy enum the rtcbundlepolicy enum defines string constants which are used to request a specific policy for gathering ice candidates if the remote peer isn't "bundle-aware" (compatible with the sdp bundle standard for bundling multiple media streams on a single transport link). all browser implementations are bundle-aware. if the remote endpoint is bundle-aware, all media tracks and data channels are bundled onto a single transport at the completion of negotiation, regardless of policy used, and any superfluous transports that were created initially are closed at that point. note: in technical terms, a bundle lets all media flow between two peers flow across a...
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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. it is used as the return value from rtcicetransport.getselectedcandidatepair() to identify the currently-selected candidate pair identified by the ice agent. 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. ...
...ent. 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. example in this example, an event handler for selectedcandidatepairchange is set up to update an on-screen display showing the protocol used by the currently selected candidate pair. ...
...tcicecandidate describing the configuration of the local end of the connection. remote the rtcicecandidate describing the configuration of the remote end of the connection. example in this example, an event handler for selectedcandidatepairchange is set up to update an on-screen display showing the protocol used by the currently selected candidate pair. ...
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RTCIceCandidatePairStats - Web APIs
tats and inherits its properties. in addition, it adds the following new properties: availableincomingbitrate optional provides an informative value representing the available inbound capacity of the network by reporting the total number of bits per second available for all of the candidate pair's incoming rtp streams. this does not take into account the size of the ip overhead, nor any other transport layers such as tcp or udp. availableoutgoingbitrate optional provides an informative value representing the available outbound capacity of the network by reporting the total number of bits per second available for all of the candidate pair's outoing rtp streams. this does not take into account the size of the ip overhead, nor any other transport layers such as tcp or udp. bytesreceieved option...
...ates. totalroundtriptime optional a floating-point value indicating the total time, in seconds, that has elapsed between sending stun requests and receiving responses to them, for all such requests made to date on this candidate pair. this includes botyh connectivity check and consent check requests. you can compute the average round trip time (rtt) by dividing this value by responsesreceived. transportid optional a domstring that uniquely identifies the rtcicetransport that was inspected to obtain the transport-related statistics (as found in rtctransportstats) used in generating this object. obsolete properties the following properties have been removed from the specification and should no longer be used. you should update any existing code to avoid using them as soon as is practical. chec...
... optional a boolean value indicating whether or not data can be received on the connection described by the candidate pair. non-standard properties selected optional a firefox-specific boolean value which is true if the candidate pair described by this object is the one currently in use. the spec-compliant way to determine the selected candidate pair is to look for a stats object of type transport, which is an rtctransportstats object. that object's selectedcandidatepairid property indicates whether or not the specified transport is the one being used. usage notes the currently-active ice candidate pair—if any—can be obtained by calling the rtcicetransport method getselectedcandidatepair(), which returns an rtcicecandidatepair object, or null if there isn't a pair selected. the acti...
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An overview of HTTP - HTTP
messages (as opposed to a stream of data). the messages sent by the client, usually a web browser, are called requests and the messages sent by the server as an answer are called responses. designed in the early 1990s, http is an extensible protocol which has evolved over time. 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. due to its extensibility, it is used to not only fetch hypertext documents, but also images and videos or to post content to servers, like with html form results. http can also be used to fetch parts of documents to update web pages on demand. components of http-based systems http is a client-server protocol: requests are sent by one entity, the user-agent ...
...alled the response. between the client and the server there are numerous entities, collectively called proxies, which perform different operations and act as gateways or caches, for example. in reality, there are more computers between a browser and the server handling the request: there are routers, modems, and more. thanks to the layered design of the web, these are hidden in the network and transport layers. http is on top, at the application layer. although important to diagnose network problems, the underlying layers are mostly irrelevant to the description of http. client: the user-agent the user-agent is any tool that acts on the behalf of the user. this role is primarily performed by the web browser; other possibilities are programs used by engineers and web developers to debug their a...
... document on demand. a server is not necessarily a single machine, but several server software instances can be hosted on the same machine. with http/1.1 and the host header, they may even share the same ip address. proxies between the web browser and the server, numerous computers and machines relay the http messages. due to the layered structure of the web stack, most of these operate at the transport, network or physical levels, becoming transparent at the http layer and potentially making a significant impact on performance. those operating at the application layers are generally called proxies. these can be transparent, forwarding on the requests they receive without altering them in any way, or non-transparent, in which case they will change the request in some way before passing it along ...
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nss tech note5
ap</big>. note that the wrappingsymkey and the tobewrappedsymkey must be on the slot where the wrap is going to happen. to move keys to the desired slot, see section moving a key from one slot to another <big>secstatus s = pk11_wrapsymkey(wrapmech, secparam, wrappingsymkey, tobewrappedsymkey, &wrappedkey);</big> <big><big>transport/store or do whatever with the wrapped key (wrappedkey.data, wrappedkey.len)</big></big> <big><big>unwrapping. </big></big> <big><big>set up the args to the function pk11_unwrapsymkey, most of which are illustrated above. the keytypemech arg of type ck_mechanism_type <big>indicates the type of key that was wrapped and can be same as the wrapmech (e.g. wrapmech=ckm_skipjack_wrap, keyt...
... that the wrappingsymkey and the tobewrappedpvtkey must be on the slot where the wrap is going to happen. to move keys to the desired slot, see section moving a key from one slot to another <big>secstatus s = pk11_wrapprivkey(slot, wrappingsymkey, tobewrappedpvtkey, wrapmech, secparam, &wrappedkey, null);</big> <big><big>transport/store or do whatever with the wrapped key (wrappedkey.data, wrappedkey.len)</big></big> <big><big>unwrapping.</big></big> prepare the args for the unwrap function. most of the args are illustrated above secitem label; /* empty, doesn't need to be freed */ label.data = null; label.len = 0; secitem *pubvalue = null; pubvalue = /* ?? */; prbool to...
... choose a wrapping mechanism. see wrapmechanismlist in security/nss/lib/pk11wrap/pk11slot.c and security/nss/lib/ssl/ssl3con.c for examples of wrapping mechanisms. most of them are cipher mechanisms. ck_mechanism_type wrapmech = ckm_des3_ecb; <big>(for example)</big> slot on which to to do th...
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nsIHttpActivityObserver
1.0 66 introduced gecko 1.8 inherits from: nsisupports last changed in gecko 1.8 (firefox 1.5 / thunderbird 1.5 / seamonkey 1.0) method overview void observeactivity(in nsisupports ahttpchannel, in pruint32 aactivitytype, in pruint32 aactivitysubtype, in prtime atimestamp, in pruint64 aextrasizedata, in acstring aextrastringdata); attributes attribute type description isactive boolean true when the interface is active and should observe http activity, otherwise false. ...
... attributes attribute type description isactive boolean true when the interface is active and should observe http activity, otherwise false. if this is false, the observeactivity() method should not be called. read only. note: this attribute is present only for compatibility and should not be used. constants activity type constants constant value description activity_type_socket_transport 0x0001 socket transport activity has occurred. activity_type_http_transaction 0x0002 http transport activity has occurred. activity subtype constants constant value description activity_subtype_request_header 0x5001 the http request is about to be queued for sending. observers can look at request headers in aextrastringdata activity_subtype_request_body_sent 0x5002 the http request's...
...ctivity_subtype_response_start 0x5003 the http response has started to arrive. activity_subtype_response_header 0x5004 the http response header has arrived. activity_subtype_response_complete 0x5005 the complete http response has been received. activity_subtype_transaction_close 0x5006 the http transaction has been closed. methods observeactivity() called when activity occurs on the http transport. you should implement this method to perform whatever tasks you wish to perform when http activity occurs. void observeactivity( in nsisupports ahttpchannel, in pruint32 aactivitytype, in pruint32 aactivitysubtype, in prtime atimestamp, in pruint64 aextrasizedata, in acstring aextrastringdata ); parameters ahttpchannel the nsihttpchannel on which the activity occurred. aactivitytype...
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RTCPeerConnection.connectionState - Web APIs
tcpeerconnection.connectionstate; value the current state of the connection, as a value from the enum rtcpeerconnectionstate. rtcpeerconnectionstate enum the rtcpeerconnectionstate enum defines string constants which describe states in which the rtcpeerconnection may be. these values are returned by the connectionstate property. ...
...alue the current state of the connection, as a value from the enum rtcpeerconnectionstate. rtcpeerconnectionstate enum the rtcpeerconnectionstate enum defines string constants which describe states in which the rtcpeerconnection may be. these values are returned by the connectionstate property. this state essentially represents the aggregate state of all ice transports (which are of type rtcicetransport or rtcdtlstransport) being used by the connection. constant description "new" at least one of the connection's ice transports (rtcicetransports or rtcdtlstransports) are in the "new" state, and none of them are in one of the following states: "connecting", "checking", "failed", or "disconnected", or all of the connection's transports are in the "closed" state. ...
...onnection. constant description "new" at least one of the connection's ice transports (rtcicetransports or rtcdtlstransports) are in the "new" state, and none of them are in one of the following states: "connecting", "checking", "failed", or "disconnected", or all of the connection's transports are in the "closed" state. "connecting" one or more of the ice transports are currently in the process of establishing a connection; that is, their rtciceconnectionstate is either "checking" or "connected", and no transports are in the "failed" state. ...
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RTCStatsReport - Web APIs
ct contains, taken from the enum type rtcstatstype. the statistic categories the type gives the name of the statistic category represented by the object, and is how you locate the specific type of data you need. the statistic category names are members of the enumerated type rtcstatstype, as follows: candidate-pair an rtcicecandidatepairstats object providing statistics related to an rtcicetransport. 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. the active candidate pair is deleted after the transport switches to another candidate pair; this change cannot be detected otherwise. certificate an rtccertificatestats object providing statistics related to...
... codec an rtccodecstats object containing statistics about a codec currently being used by rtp streams to send or receive data for the rtcpeerconnection. csrc an rtccontributingsourcestats object which contains statistics related to a contributing source (csrc) that contributed to an inbound rtp stream. ...
...since this only provides statistics related to inbound data, without considering the local peer's state, any values that require knowledge of both, such as round-trip time, is not included. this report isn't available if there are no connected peers. local-candidate an rtcicecandidatestats object giving statistics about an ice local candidate; these candidates are found in the output from rtcicetransport.getlocalcandidates(). outbound-rtp the report is an rtcoutboundrtpstreamstats object providing statistics based on the local peer's outgoing data being sent to its peers. this information considers only the outbound rtp stream, so any data which requires information about the state of the remote peers (such as round-trip time) is unavailable, since those values can't be computed without knowing...
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WebRTC Statistics API - Web APIs
ng dictionaries; for each statistic category, the full hierarchy of rtcstats-based dictionaries are listed, so you can easily find all the available values. mapping of statistic category names to the dictionaries they implement statistic category name (rtcstatstype) description dictionaries implemented candidate-pair statistics describing the change from one rtcicetransport to another, such as during an ice restart. rtcicecandidatepairstats rtcstats certificate statistics about a certificate being used by an rtcicetransport. rtccertificatestats rtcstats codec statistics about a specific codec being used by streams being sent or received by th...
...tp statistics describing the state of one of the connection's inbound data streams. rtcinboundrtpstreamstats rtcreceivedrtpstreamstats rtcrtpstreamstats rtcstats local-candidate statistics about a local ice candidate associated with the connection's rtcicetransports. rtcicecandidatestats rtcstats media-source statistics about the media produced by the mediastreamtrack attached to an rtp sender. the dictionary this key maps to depends on the track's kind. rtcaudiosourcestats or rtcvideosourcestats rtcmediasourcestats rtcstats ...
...he statistics depends on the media kind. rtcaudioreceiverstats or rtcvideoreceiverstats rtcaudiohandlerstats or rtcvideohandlerstats rtcmediahandlerstats rtcstats remote-candidate statistics about a remote ice candidate associated with the connection's rtcicetransports. rtcicecandidatestats rtcstats remote-inbound-rtp statistics describing the state of the inbound data stream from the perspective of the remote peer. rtcremoteinboundrtpstreamstats rtcreceivedrtpstreamstats rtcrtpstreamstats rtcstats ...
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HSTS - MDN Web Docs Glossary: Definitions of Web-related terms
http strict transport security lets a web site inform the browser that it should never load the site using http and should automatically convert all attempts to access the site using http to https requests instead. it consists in one http header, strict-transport-security, sent by the server with the resource. in other words, it tells the browser that changing the protocol from http to https in a url works (and is more secure) and asks the browser to do it for every request. ...
...g http and should automatically convert all attempts to access the site using http to https requests instead. it consists in one http header, strict-transport-security, sent by the server with the resource. in other words, it tells the browser that changing the protocol from http to https in a url works (and is more secure) and asks the browser to do it for every request. ...
...empts to access the site using http to https requests instead. it consists in one http header, strict-transport-security, sent by the server with the resource. in other words, it tells the browser that changing the protocol from http to https in a url works (and is more secure) and asks the browser to do it for every request. ...
...s requests instead. it consists in one http header, strict-transport-security, sent by the server with the resource. in other words, it tells the browser that changing the protocol from http to https in a url works (and is more secure) and asks the browser to do it for every request. ...

Multithreading in Necko
protocol handlers (eg. file, finger, datetime) do not worry about threading issues. the internal i/o interfaces default to proxying all callbacks to the main thread. this keeps simple protocol handlers simple :-) more complex protocol handlers (eg. http, ftp) operate partially on the background threads to improve performance. for example, the http protocol handler has code that runs on the socket transport thread to kick off pending requests well before the main thread would get around to doing so. socket transport thread (1) socket connections are processed on a single background thread. this thread is controlled by the nssockettransportservice. it sits in a loop, polling a list of sockets. when a socket can be read, the socket's listener is notified either synchronously (on the same thread) or...
...thread would get around to doing so. socket transport thread (1) socket connections are processed on a single background thread. this thread is controlled by the nssockettransportservice. it sits in a loop, polling a list of sockets. when a socket can be read, the socket's listener is notified either synchronously (on the same thread) or asynchronously via a nsistreamlistenerproxy impl. file transport threads (1-4) file i/o is processed via a thread pool that has between 1 and 4 threads. these threads are controlled by the nsfiletransportservice. each file transport object is given time on one of the file transport threads. during which it reads (writes) until its buffers are full (empty). once at the limit of its buffers, it yields the thread to another pending file transport object. as with...
...4) file i/o is processed via a thread pool that has between 1 and 4 threads. these threads are controlled by the nsfiletransportservice. each file transport object is given time on one of the file transport threads. during which it reads (writes) until its buffers are full (empty). once at the limit of its buffers, it yields the thread to another pending file transport object. as with the socket transport thread, the nsistreamlistener passed to a file transport's asyncread method can operate partially on the file transport's thread before proxying data to the main thread. the file transport is confusingly named since it is not restricted to loading files. it simply promises to read from a nsiinputstream (or write to a nsioutputstream) on a background thread. the implementation of the stream should...
...the jar protocol handler, for rexample, passes a special nsiinputstream impl that does gzip decoding on the fly. dns thread (0-1) on most platforms dns requests a processed on a background thread. however, the details of the implementation are heavily platform dependent. ...

nsIScriptableIO
in nsivariant abase, in astring amode, [optional] in astring acharset, [optional] in astring areplacechar, [optional] in unsigned long abuffersize ); parameters abase the base object from which to read. this should be one of the following types: nsifile: an object returned by getfile() or getfilewithpath(), or any object implementing the nsifile interface. nsitransport: a transport object such as a socket. nsiinputstream: a stream returned by a previous call to this method, or any other object that implements nsiinputstream. string: a string from which to read data. amode a space-delineated string of mode terms controlling the manner in which the stream is accessed. if none of these modes are desired, specify an empty string. text: read ...
...hold a block of the next piece of the data to be read. this mode is generally used as a wrapper for other streams. by default, the buffer size is 1024 bytes; however, the size may be changed by specifying the abuffersize parameter. if the text mode is specified, a buffer is always used, regardless of whether or not it is specifically requested. block: when this mode is specified, reading from a transport such as a socket when no data is available will block until data is available before returning. if this mode isn't specified, an exception is thrown if no data is available. deleteonclose: the file is automatically deleted when the stream is closed. this can be useful for temporary files. closeoneof: the file is automatically closed when the end of the file is reached. reopenonrewind: used ...
... [optional] in astring acharset, [optional] in astring areplacechar, [optional] in unsigned long abuffersize, [optional] in unsigned long apermissions ); parameters abase the base object to which to write. this should be one of the following types: nsifile: an object returned by getfile() or getfilewithpath(), or any object implementing the nsifile interface. nsitransport: a transport object such as a socket. nsioutputstream: a stream returned by a previous call to this method, or any other object that implements nsioutputstream. amode a space-delineated string of mode terms controlling the manner in which the stream is accessed. if none of these modes are desired, specify an empty string. text: write unicode-converted text. the default chara...
...gardless of whether or not it is specifically requested. append: when writing to files, indicates to append to the file rather than overwrite the existing file. the write position will be set to the end of the file before each write. if used in conjuction with the create mode, an existing file may be opened for appending, or a new file created. block: when this mode is specified, writing to a transport such as a socket doesn't return until all data has been written. this may cause a delay if the socket's underlying buffer is full. if you don't specify this mode, an exception is thrown if the buffer is full. nocreate: if the file doesn't already exist, don't create a new file. notruncate: if the file already exists, overwrite the existing content instead of resetting the file size to 0 bytes...

RTCIceCandidate.foundation - Web APIs
note: if port is null — and port is supported by the user agent — passing the candidate to addicecandidate() will fail, throwing an operationerror exception. ...
... note: if port is null — and port is supported by the user agent — passing the candidate to addicecandidate() will fail, throwing an operationerror exception. ...
... note: if port is null — and port is supported by the user agent — passing the candidate to addicecandidate() will fail, throwing an operationerror exception. ...
...-line) which describes an ice candidate: a=candidate:4234997325 1 udp 2043278322 192.168.0.56 44323 typ host the field "4234997325" is the foundation. example this code snippet uses the foundation of two candidates to determine if they're actually the same candidate. if (candidate1.foundation == candidate2.foundation) { /* the two candidates are the same, even if they're on different transports */ } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecandidate.foundation' in that specification. candidate recommendation initial definition. ...

RTCIceCandidatePairStats.selected - Web APIs
r or not the candidate pair described by the object is the one currently being used to communicate with the remote peer. syntax isselected = icpstats.selected; value a firefox-specific boolean value which is true if the candidate pair described by this object is the one currently in use. ...
...r described by the object is the one currently being used to communicate with the remote peer. syntax isselected = icpstats.selected; value a firefox-specific boolean value which is true if the candidate pair described by this object is the one currently in use. in any other browser, you can determine the selected candidate pair by looking for a stats object of type transport, which is an rtctransportstats object. ...
...ected = icpstats.selected; value a firefox-specific boolean value which is true if the candidate pair described by this object is the one currently in use. in any other browser, you can determine the selected candidate pair by looking for a stats object of type transport, which is an rtctransportstats object. that object's selectedcandidatepairid property indicates whether or not the specified transport is the one being used. 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. if that fails, then the second section iterates over the repo...
...id is used to get the rtcicecandidatepair describing the connection. if that fails, then the second section iterates over the reports, looking for a candidate-pair record whose firefox-specific selected property is true. this candidate pair is then returned as the currently-selected one. function getcurrentcandidatepair(statsresults) { statsresults.foreach(report => { if (report.type === "transport") { currentpair = statsresults.get(report.selectedcandidatepairid); } }); if (!currentpair) { statsresults.foreach(report => { if (report.type === "candidate-pair" && report.selected) { currentpair = report; } }); } return currentpair; } specifications not part of any specification. this property is unique to firefox. ...

RTCStatsType - Web APIs
the rtcstatstype enumerated type (enum) is a set of strings which define the type of statistics reported in a record found in the rtcstatsreport object. this type determines which of the rtcstats-based dictionaries the record is based upon. values candidate-pair an rtcicecandidatepairstats object providing statistics related to an rtcicetransport. 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. the active candidate pair is deleted after the transport switches to another candidate pair; this change cannot be detected otherwise. certificate an rtccertificatestats object providing statistics related to a certificate being used by an rtcicetransport. codec...
...since this only provides statistics related to inbound data, without considering the local peer's state, any values that require knowledge of both, such as round-trip time, is not included. this report isn't available if there are no connected peers. local-candidate an rtcicecandidatestats object giving statistics about an ice local candidate; these candidates are found in the output from rtcicetransport.getlocalcandidates(). outbound-rtp the report is an rtcoutboundrtpstreamstats object providing statistics based on the local peer's outgoing data being sent to its peers. this information considers only the outbound rtp stream, so any data which requires information about the state of the remote peers (such as round-trip time) is unavailable, since those values can't be computed without knowing...
...des statistics related to the overall peer connection's rtcpeerconnection. receiver provides statistics about a specific rtcrtpreceiver. the statistics object is an rtcaudioreceiverstats object if kind is audio; if kind is video, the object is an rtcvideoreceiverstats object. remote-candidate the report is an rtcicecandidatestats object containing statistics about the remote candidate's rtcicetransport. this may include information such as the type of network, the protocol, the url, the type of relay being used, and so forth. remote-inbound-rtp the report is an rtcremoteinboundrtpstreamstats object providing statistics about your outbound rtp data stream, but from the perspective of the remote peer. that is, this information is about your outbound-rtp stream, but as seen by the remote device ...
...connection. track the object is one of the types based on rtcmediahandlerstats: for audio tracks, the type is rtcsenderaudiotrackattachmentstats and for video tracks, the type is rtcsendervideotrackattachmentstats. the data within provides statistics related to a particular mediastreamtrack's attachment to an rtcrtpsender; also included are the media level metrics that go along with the track. transport an object that contains statistics related to a transport for an rtcpeerconnection. the object is of type rtctransportstats. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcstatstype' in that specification. candidate recommendation initial definition. ...

Signaling and video calling - Web APIs
gnaling server to resolve how to connect them over the internet. a signaling server's job is to serve as an intermediary to let two peers find and establish a connection while minimizing exposure of potentially private information as much as possible. how do we create this server and how does the signaling process actually work? first we need the signaling server itself. webrtc doesn't specify a transport mechanism for the signaling information. you can use anything you like, from websocket to xmlhttprequest to carrier pigeons to exchange the signaling information between the two peers. it's important to note that the server doesn't need to understand or interpret the signaling data content. although it's sdp, even this doesn't matter so much: the content of the message going through the signalin...
...he setlocaldescription() fulfillment handler has run, the browser begins firing icecandidate events that the callee must handle, one for each candidate that needs to be transmitted to the remote peer. sending ice candidates the ice negotiation process involves each peer sending candidates to the other, repeatedly, until it runs out of potential ways it can support the rtcpeerconnection's media transport needs. since ice doesn't know about your signaling server, your code handles transmission of each candidate in your handler for the icecandidate event. your onicecandidate handler receives an event whose candidate property is the sdp describing the candidate (or is null to indicate that the ice layer has run out of potential configurations to suggest). the contents of candidate are what you need...
...is function constructs an rtcicecandidate object by passing the received sdp into its constructor, then delivers the candidate to the ice layer by passing it into mypeerconnection.addicecandidate(). this hands the fresh ice candidate to the local ice layer, and finally, our role in the process of handling this candidate is complete. each peer sends to the other peer a candidate for each possible transport configuration that it believes might be viable for the media being exchanged. at some point, the two peers agree that a given candidate is a good choice and they open the connection and begin to share media. it's important to note, however, that ice negotiation does not stop once media is flowing. instead, candidates may still keep being exchanged after the conversation has begun, either while tr...
...t some point, the two peers agree that a given candidate is a good choice and they open the connection and begin to share media. it's important to note, however, that ice negotiation does not stop once media is flowing. instead, candidates may still keep being exchanged after the conversation has begun, either while trying to find a better connection method, or simply because they were already in transport when the peers successfully established their connection. in addition, if something happens to cause a change in the streaming scenario, negotiation will begin again, with the negotiationneeded event being sent to the rtcpeerconnection, and the entire process starts again as described before. this can happen in a variety of situations, including: changes in the network status, such as a bandw...

Using WebRTC data channels - Web APIs
connection. but there's a lot more you can do with webrtc. 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. since all webrtc components are required to use encryption, any data transmitted on an rtcdatachannel is automatically secured using datagram transport layer security (dtls). see security below for more information. creating a data channel the underlying data transport used by the rtcdatachannel can be created in one of two ways: let webrtc create the transport and announce it to the remote peer for you (by causing it to receive a datachannel event). this is the easy way, and works for a wide variety of use cases, but may not be flexible e...
... let's look at each of these cases, starting with the first, which is the most common. automatic negotiation often, you can allow the peer connection to handle negotiating the rtcdatachannel connection for you. ...
...channel completely. <<<write more about using bufferedamount, bufferedamountlowthreshold, onbufferedamountlow, and bufferedamountlow here>>> ... understanding message size limits for any data being transmitted over a network, there are size restrictions. at a fundamental level, the individual network packets can't be larger than a certain value (the exact number depends on the network and the transport layer being used). at the application level—that is, within the user agent's implementation of webrtc on which your code is running—the webrtc implementation implements features to support messages that are larger than the maximum packet size on the network's transport layer. this can complicate things, since you don't necessarily know what the size limits are for various user agents, and ho...
... written by lennart grahl. he goes into a bit more detail there, but as browsers have been updated since then some of it may be out-of-date. in addition, as time goes by, it will become more so, especially once eor and ndata support are fully integrated in the major browsers. security all data transferred using webrtc is encrypted. in the case of rtcdatachannel, the encryption used is datagram transport layer security (dtls), which is based on transport layer security (tls). since tls is used to secure every https connection, any data you send on a data channel is as secure as any other data sent or received by the user's browser. more fundamentally, since webrtc is a peer-to-peer connection between two user agents, the data never passes through the web or application server. this reduces oppor...

MDN Web Docs Glossary: Definitions of Web-related terms
erialization developer tools dhtml digest digital certificate distributed denial of service dmz dns doctype document directive 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 ...
...n rendering engine repo reporting directive request header resource timing response header responsive web design rest rgb ril robots.txt round trip time (rtt) routers rss rtcp (rtp control protocol) rtf rtl (right to left) rtp (real-time transport protocol) and srtp (secure rtp) rtsp: real-time streaming protocol ruby s safe same-origin policy scm scope screen reader script-supporting element scroll container scrollport sctp sdp search engine second-level domain secure sockets layer (ssl) ...
...key cryptography synchronous syntax syntax error synthetic monitoring t tag tcp tcp handshake tcp slow start telnet texel thread three js time to first byte time to interactive tld tofu transmission control protocol (tcp) transport layer security (tls) tree shaking trident truthy ttl turn type type coercion type conversion u udp (user datagram protocol) ui undefined unicode uri url urn usenet user agent utf-8 ux v v...

Index
password based encryption (pbe). this is often sufficient if you encrypt data for yourself, but as soon as you need to exchange signed/encrypted data with communication partners, using public key encryption simplifies the key management. the environment that describes how to use public key encryption is called public key infrastructure (pki). the public keys that are exchanged between parties are transported using a container; the container is called a certificate, following standard x.509 version 3. a certificate contains lots of other details; for example, it contains a signature by a third party that expresses trust in the ownership relationship for the certificate. the trust assigned by the third party might be restricted to certain uses, which are listed in certificate extensions that are cont...
...sing a different one, without restarting. such an operation might fail at runtime if data elements are still being referenced. in addition to the freebl, softoken, and ckbi modules, there is an utility library for general operations (e.g., encoding/decoding between data formats, a list of standardized object identifiers (oid)). nss has an ssl/tls module that implements the secure sockets layer/transport layer security network protocols, an s/mime module that implements cms messaging used by secure email and some instant messaging implementations, a dbm library that implements the classic database storage, and finally a core nss library for the big set of “everything else”. newer generations of the database use the sqlite database to allow concurrent access by multiple applications. all of...
...es it to another file, this illustrates the use of nss message apis. 175 initialize nss database - sample 2 html, javascript, nss, nss article, nss initialization, web development the nss sample code below demonstrates how to initialize the nss database. 176 nss sample code sample1 this is an example program that demonstrates how to do key generation and transport between cooperating servers. this program shows the following: 177 nss sample code sample_1_hashing examples, nss, security this is an example program that demonstrates how to compute the hash of a file and save it to another file. this program illustrates the use of nss message apis. 178 nss sample code sample_2_initialization of nss examples, nss, security ...

nsIHttpActivityDistributor
this is primarily useful for debuggers and other traffic monitoring tasks. ...
...void removeobserver( in nsihttp...
...tpactivitydistributor); method overview void addobserver(in nsihttpactivityobserver aobserver); void removeobserver(in nsihttpactivityobserver aobserver); methods addobserver() begins delivery of notifications of http transport activity. void addobserver( in nsihttpactivityobserver aobserver ); parameters aobserver the nsihttpactivityobserver that should receive notifications of http transport activity; this object's nsihttpactivityobserver.observeactivity() method will be called each time activity occurs. removeobserver() stops delivery of notifications of http transport activity. void removeobserver( in nsihttpactivityobserver aobserver ); parameters aobserver the nsihttpactivityobserver that should no longer receive notifications of http transport activity. see also monitorin...

nsIServerSocketListener
netwerk/base/public/nsiserversocket.idlscriptable this interface is notified whenever a server socket accepts a new connection. the transport is in the connected state, and read/write streams can be opened using the normal nsitransport api. the address of the client can be found by calling the nsisockettransport.getaddress() method or by inspecting nsisockettransport.gethost(), which returns a string representation of the client's ip address, which may be either an ipv4 or an ipv6 address. ...
...i. the address of the client can be found by calling the nsisockettransport.getaddress() method or by inspecting nsisockettransport.gethost(), which returns a string representation of the client's ip address, which may be either an ipv4 or an ipv6 address. 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. void onsocketaccepted( in nsiserversocket aserv, in nsisockettransport atransport ); parameters aserv the server socket. atransport the connected socket transport. onstoplistening() this metho...
... stops for some reason. ...

nsISocketProvider
ort the port for this connection. aproxyhost if non-null, the proxy hostname for this connection. aproxyport the proxy port for this connection. aflags control flags that govern this connection (see constants). afiledesc the existing prfiledesc. asecurityinfo any security info that should be associated with afiledesc. this object typically implements nsitransportsecurityinfo. native code only!newsocket creates a new socket. void newsocket( in long afamily, in string ahost, in long aport, in string aproxyhost, in long aproxyport, in unsigned long aflags, out prfiledescstar afiledesc, out nsisupports asecurityinfo ); parameters afamily the address family for this socket (pr_af_inet or pr_af_inet6). ahost the hostname for t...
...ort the port for this connection. ...
... aproxyhost if non-null, the proxy hostname for this connection. ...

PerformanceTiming - Web APIs
he same as performancetiming.fetchstart. performancetiming.domainlookupend read only when the domain lookup is finished. 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. ...
...if a persistent connection is used, the value will be the same as performancetiming.fetchstart. ...
...nsidered as opened when all secure connection handshake, or socks authentication, is terminated. performancetiming.secureconnectionstart read only when the secure connection handshake starts. if no such connection is requested, it returns 0. performancetiming.requeststart read only when the browser sent the request to obtain the actual document, from the server or from a cache. 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.responsestart read only when the browser received the first byte of the response, from the server from a cache, or from a local resource. performancetiming.responseend read only when the browser received the last byte of t...

PublicKeyCredentialRequestOptions.allowCredentials - Web APIs
lue an array whose elements are objects with the following properties: type a string describing type of public-key credential to be created. as of this writing (march 2019), only "public-key" may be used. id a buffersource matching an existing public key credential identifier (publickeycredential.rawid). this identifier is generated during the creation of the publickeycredential instance. transports optional an array of strings describing the possible transports between the client and the authenticator. the value of the strings may be: "usb": the authenticator can be contacted via a removable usb link "nfc": the authenticator may be used over nfc (near field communication) "ble": the authenticator may be used over ble (bluetooth low energy) "internal": the authenticator is specif...
...ver nfc (near field communication) "ble": the authenticator may be used over ble (bluetooth low energy) "internal": the authenticator is specifically bound to the client device (cannot be removed). if the authenticator does not contain any of these public key credentials, the client will throw a domexception "notallowederror". ...
...ticator is specifically bound to the client device (cannot be removed). if the authenticator does not contain any of these public key credentials, the client will throw a domexception "notallowederror". ...

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. syntax var state = adatachannel.readystate; values a string which is one of the values in the rtcdatachannelstate enum, indicating the current state of the underlying data transport. 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. this is t...
... 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. this is the state of a new rtcdatachannel after being created by rtcpeerconnection.createdatachannel() (on the peer which started the connection process). "open" the underlying data transport has been established and data can be transferred bidirectionally across it. this is the default state of a new rtcdatachannel created by the webrtc layer when the remote peer created the channel and delivered to the site or app in a datachannel event of type rtcdatachannelevent. "closing" the process of closing the underlying data transport has begun. it is no longer possible to queue...
...remote peer created the channel and delivered to the site or app in a datachannel event of type rtcdatachannelevent. "closing" the process of closing the underlying data transport has begun. it is no longer possible to queue new messages to be sent, but previously queued messages may still be send or received before entering the "closed" state. "closed" the underlying data transport has closed, or the attempt to make the connection failed. example var datachannel = peerconnection.createdatachannel("file transfer"); var sendqueue = []; function sendmessage(msg) { switch(datachannel.readystate) { case "connecting": console.log("connection not open; queueing: " + msg); sendqueue.push(msg); break; case "open": sendqueue.foreach((msg) =>...

RTCDataChannel - Web APIs
ed down.onclosing the rtcdatachannel.onclosing property is an eventhandler which specifies a function to be called by the browser when the closing event is received by the rtcdatachannel. this is a simple event which indicates that the data channel is being closed, that is, rtcdatachannel transitions to "closing" state. for example, after rtcdatachannel.close() was called but the underlying data transport might not have been closed yet.onerror the rtcdatachannel.onerror property is an eventhandler which specifies a function to be called when the error event is received. when an error occurs on the data channel, the function receives as input an errorevent object describing the error which occurred.onmessage the rtcdatachannel.onmessage property stores an eventhandler which specifies a function t...
...lled when the message event is fired on the channel. this event is represented by the messageevent interface. this event is sent to the channel when a message is received from the other peer.onopen the rtcdatachannel.onopen property is an eventhandler which specifies a function to be called when the open event is fired; this is a simple event which is sent when the data channel's underlying data transport—the link over which the rtcdatachannel's messages flow—is established or re-established.methodsalso inherits methods from: eventtargetclose()the rtcdatachannel.close() method closes the rtcdatachannel. either peer is permitted to call this method to initiate closure of the channel.send()the send() method of the rtcdatachannel interface sends data across the data channel to the remote peer. e...
...the channel.send()the send() method of the rtcdatachannel interface sends data across the data channel to the remote peer. events bufferedamountlow sent to the channel's onbufferedamountlow event handler when the number of bytes of data in the outgoing data buffer falls below the value specified by bufferedamountlowthreshold. close sent to the onclose event handler when the underlying data transport closes. error sent to the onerror event handler when an error occurs on the data channel. message sent to the onmessage event handler when a message has been received from the remote peer. the message contents can be found in the event's data property. open sent to the onopen event handler when the data channel is first opened, or when an existing data channel's underlying connection re-ope...

RTCIceCandidate.protocol - Web APIs
the rtcicecandidate interface's read-only protocol property is a string which indicates whether the candidate uses udp or tcp as its transport protocol. 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(). the protocol property's value is set when the rtcicecandidate() constructor is used. the value is automatically extracted from the candidate a-line, if it's formatted properly. syntax var protocol = rtcicecandidate.protocol; value a domstring which indicates what network protocol the candidate uses, udp or tcp. ...
...hen the rtcicecandidate() constructor is used. the value is automatically extracted from the candidate a-line, if it's formatted properly. syntax var protocol = rtcicecandidate.protocol; value a domstring which indicates what network protocol the candidate uses, udp or tcp. these values are defined by the enumerated type rtciceprotocol: tcp the candidate, if selected, would use tcp as the transport protocol for its data. the tcptype property provides additional information about the kind of tcp candidate represented by the object. udp the candidate will use the udp transport protocol for its data. this is the preferred protocol for media interactions because of its better performance profile. note: if protocol is null — and protocol is supported by the user agent — passing the cand...
...protocol is null — and protocol is supported by the user agent — passing the candidate to addicecandidate() will fail, throwing an operationerror exception. usage notes here's an example candidate a-line from an ice transaction: a=candidate:4234997325 1 udp 2043278322 192.168.0.56 44323 typ host the third field, "udp", is the protocol type, indicating that the candidate would use the udp transport protocol. example this code snippet examines the value of protocol to decide if it should look at the value of tcptype to see if it's a simultaneous-open (s-o) candidate. if (candidate.protocol == "tcp") { if (candidate.tcptype == "so") { adjustforsimultaneousopen(candidate); } specifications specification status comment webrtc 1.0: real-time communication betw...

RTCIceProtocol - Web APIs
this is the preferred protocol for media interactions because of its better performance profile. ...
...this is the preferred protocol for media interactions because of its better performance profile. ...
...this is the preferred protocol for media interactions because of its better performance profile. usage notes the rtciceprotocol type is used by the protocol property of rtcicecandidate objects. ...

RTCIceTcpCandidateType - Web APIs
usage notes the rtcicetcpcandidatetype type is used by the tcptype property of rtcicecandidate objects. ...
... usage notes the rtcicetcpcandidatetype type is used by the tcptype property of rtcicecandidate objects. ...
... usage notes the rtcicetcpcandidatetype type is used by the tcptype property of rtcicecandidate objects. ...

RTCRtpSender - Web APIs
which can be used to send dtmf tones using telephone-event payloads on the rtp session represented by the rtcrtpsender object. if null, the track and/or the connection doesn't support dtmf. only audio tracks can support dtmf. rtcrtpsender.track read only the mediastreamtrack which is being handled by the rtcrtpsender. if track is null, the rtcrtpsender doesn't transmit anything. ...
...ones using telephone-event payloads on the rtp session represented by the rtcrtpsender object. if null, the track and/or the connection doesn't support dtmf. only audio tracks can support dtmf. rtcrtpsender.track read only the mediastreamtrack which is being handled by the rtcrtpsender. if track is null, the rtcrtpsender doesn't transmit anything. ...
...f. only audio tracks can support dtmf. rtcrtpsender.track read only the mediastreamtrack which is being handled by the rtcrtpsender. if track is null, the rtcrtpsender doesn't transmit anything. rtcrtpsender.transport read only the rtcdtlstransport over which the sender is exchanging the rtp and rtcp packets used to manage transmission of media and control data. this value is null until the transport is established. when bundling is in use, more than transceiver may be sharing the same transport object. obsolete properties rtcptransport this property has been removed; the rtp and rtcp transports have been combined into a single transport. use the transport property instead. methods rtcrtpsender.getparameters() returns a rtcrtpparameters object describing the current configurat...

HTTP headers - HTTP
rization te trailer transfer-encoding upgrade (see also protocol upgrade mechanism). end-to-end headers these headers must be transmitted to the final recipient of the message: the server for a request, or the client for a response. intermediate proxies must retransmit these headers unmodified and caches must store them. 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. authentication www-authenticate defines the authentication method that should be used to access a resource. authorization contains the credentials to authenticate a user-agent with a server. proxy-authenticate defines the authentication ...
...cate transparency requirements, which prevents the use of misissued certificates for that site from going unnoticed. when a site enables the expect-ct header, they are requesting that chrome check that any certificate for that site appears in public ct logs. feature-policy provides a mechanism to allow and deny the use of browser features in its own frame, and in iframes that it embeds. strict-transport-security (hsts) force communication using https instead of http. upgrade-insecure-requests sends a signal to the server expressing the client’s preference for an encrypted and authenticated response, and that it can successfully handle the upgrade-insecure-requests directive. x-content-type-options disables mime sniffing and forces browser to use the type given in content-type. x-download...
...esponse cycle. service-worker-allowed used to remove the path restriction by including this header in the response of the service worker script. sourcemap links generated code to a source map. upgrade the relevant rfc document for the upgrade header field is rfc 7230, section 6.7. the standard establishes rules for upgrading or changing to a different protocol on the current client, server, transport protocol connection. for example, this header standard allows a client to change from http 1.1 to http 2.0, assuming the server decides to acknowledge and implement the upgrade header field. neither party is required to accept the terms specified in the upgrade header field. it can be used in both client and server headers. if the upgrade header field is specified, then the sender must also send ...

HTTP resources and specifications - HTTP
rfc 2183 communicating presentation information in internet messages: the content-disposition header field only a subset of syntax of the content-disposition header can be used in the context of http messages. proposed standard rfc 7239 forwarded http extension proposed standard rfc 6455 the websocket protocol proposed standard rfc 5246 the transport layer security (tls) protocol version 1.2 this specification has been modified by subsequent rfcs, but these modifications have no effect on the http protocol. proposed standard rfc 8446 the transport layer security (tls) protocol version 1.3 supersedes tls 1.2. proposed standard rfc 2817 upgrading to tls within http/1.1 proposed standard rfc 7540...
...ol version 1.3 supersedes tls 1.2. proposed standard rfc 2817 upgrading to tls within http/1.1 proposed standard rfc 7540 hypertext transfer protocol version 2 (http/2) proposed standard rfc 7541 hpack: header compression for http/2 on the standard track rfc 7838 http alternative services on the standard track rfc 7301 transport layer security (tls) application-layer protocol negotiation extension used to negotiate http/2 at the transport to save an extra request/response round trip. proposed standard rfc 6454 the web origin concept proposed standard fetchthe definition of 'cors' in that specification. cross-origin resource sharing living standard rfc 7034 http header fiel...
...on used to negotiate http/2 at the transport to save an extra request/response round trip. proposed standard rfc 6454 the web origin concept proposed standard fetchthe definition of 'cors' in that specification. ...

A typical HTTP session - HTTP
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). the client sends its request, and waits for the answer. the server processes the request, sending back its answer, providing a status code and appropriate data. 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. opening a connection in http means initiating a connection in the underlying transport layer, usually this is tcp. with tcp the default p...
.... the final block is a data block, which contains the optional data. ...
...(contains a site-customized page helping the user to find the missing resource) response status codes http response status codes indicate if a specific http request has been successfully completed. respo...

Navigation and resource timings - Web Performance
rect, or if one of the redirects is not of the same origin, the value returned is 0. redirectend when the last http redirect is completed, that is when the last byte of the http response has been received. if there is no redirect, or if one of the redirects is not of the same origin, the value returned is 0. connectend when the connection is opened network. if the transport layer reports an error and the connection establishment is started again, the last connection establishment 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. connectstart when the request to open a connectio...
...onnection establishment is started again, the last connection establishment 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. connectstart when 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 establishment start time is given. if a persistent connection is used, the value will be the same as performancetiming.fetchstart. domainlookupend when the domain lookup is finished. if a persistent connection is used, or the information is stored in a cache or a local resource, the value wil...
...the information is stored in a cache or a local resource, the value will be the same as performancetiming.fetchstart. fetchstart when the browser is ready to fetch the document using an http request. this moment is before the check to any application cache. requeststart when the browser sent the request to obtain the actual document, from the server or from a cache. 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. responsestart when the browser received the first byte of the response, from the server from a cache, or from a local resource. responseend when the browser received the last byte of the response, or when the connection is close...

Updating addons broken by private browsing changes - Archive of obsolete content
e to a change described here and you are having trouble updating it. ...
...warelistener should be used instead, or no notifications will be received for private downloads. there is a new download-manager-remove-download-guid notification, which passes an nsisupportscstring subject for the guid or null. nsihttpauthmanager: setauthidentity and getauthidentity now take an optional boolean parameter to indicate whether the identity is classified as private. nsistricttransportsecurityservice: processstsheader, removestsstate, isstshost, and isstsuri now take a flags parameter that understands nsisocketprovider.no_permanent_storage or nothing. nsiprivatebrowsingservice: this service is deprecated, and as of firefox 20 it is now an empty interface. if your code depends on this interface, it will break. the global private browsing service is actively being removed, and...

Mozilla Application Framework in Detail - Archive of obsolete content
d or removed. flexible programming interfaces such as the web shell api, the java wrapper api and the gecko activex control make gecko easily embeddable into applications and devices. full browser functionality: gecko allows any desktop application as well as new devices, products and appliances to be web-enabled, with complete browser functionality. supported standards and features include data transport protocols, multilingual character data, image data, java and plugins. open source: gecko represents the first software release from netscape and mozilla.org's open source initiative. by harnessing the talent of thousands of developers worldwide, mozilla.org has accelerated internet software development and increased the quality of the final product, delivering unmatched support for web standards...
...ko browser engine or gecko technologies, including distributing gecko inside another product. furthermore, when embedding gecko, developers are free to choose what modules to include. there is no requirement for distribution of any other software. the necko network library mozilla's network library (aka necko) provides an extensible, platform-independent api for several layers of networking from transport to presentation layers. its provides a generic and extensible framework for fetching urls with support for common protocols (http, ftp, file, etc.) and a way to plug in custom protocol handlers. necko also provides an abstract "file descriptor" interface for low-level data access. necko features include support for asynchronous i/o, a generic disk and memory cache service, asynchronous caching dn...

Introduction to SSL - Archive of obsolete content
introduction this document introduces the secure sockets layer (ssl) protocol. ssl has been universally accepted on the world wide web for authenticated and encrypted communication between clients and servers. the ssl protocol ciphers used with ssl the ssl handshake the new internet engineering task force (ietf) standard protocol called transport layer security (tls) is based on ssl. the details of the protocol are available in request for comments (rfc): 2246,the tls protocol version 1.0. some red hat products already support tls. most other red hat products plan to support the protocol in future versions. this document is primarily intended for administrators of red hat server products, but the information it contains may also be useful for developers of applications that support ssl. t...
...ended for administrators of red hat server products, but the information it contains may also be useful for developers of applications that support ssl. the document assumes that you are familiar with the basic concepts of public-key cryptography, as summarized in "introduction to public-key cryptography." the ssl protocol the transmission control protocol/internet protocol (tcp/ip) governs the transport and routing of data over the internet. other protocols, such as the hypertext transport protocol (http), lightweight directory access protocol (ldap), or internet messaging access protocol (imap), run "on top of" tcp/ip in the sense that they all use tcp/ip to support typical application tasks such as displaying web pages or running email servers. the ssl protocol runs above tcp/ip and below h...

Tracking Protection
runs when content from a tracking site is loaded, then the callback will not execute. ...
... note: depending on a third party in this way is not a good practice anyway, because then your site can be broken if the third party is slow or ...

An overview of NSS Internals
password based encryption (pbe). this is often sufficient if you encrypt data for yourself, but as soon as you need to exchange signed/encrypted data with communication partners, using public key encryption simplifies the key management. the environment that describes how to use public key encryption is called public key infrastructure (pki). the public keys that are exchanged between parties are transported using a container; the container is called a certificate, following standard x.509 version 3. a certificate contains lots of other details; for example, it contains a signature by a third party that expresses trust in the ownership relationship for the certificate. the trust assigned by the third party might be restricted to certain uses, which are listed in certificate extensions that are cont...
...sing a different one, without restarting. such an operation might fail at runtime if data elements are still being referenced. in addition to the freebl, softoken, and ckbi modules, there is an utility library for general operations (e.g., encoding/decoding between data formats, a list of standardized object identifiers (oid)). nss has an ssl/tls module that implements the secure sockets layer/transport layer security network protocols, an s/mime module that implements cms messaging used by secure email and some instant messaging implementations, a dbm library that implements the classic database storage, and finally a core nss library for the big set of “everything else”. newer generations of the database use the sqlite database to allow concurrent access by multiple applications. all of...

NSS Sample Code Sample1
this program shows the following: rsa key pair generation naming rsa key pairs looking up a previously generated key pair by name creating aes and mac keys (or encryption and mac keys in general) wrapping symmetric keys using your own rsa key pair so that they can be stored on disk or in a database. ...
...eypair that will be used // store keys locally. // 1b. the primary host wraps the newly generated key using the // rsa key and stores the wrapped key data in a local file. // 1c. the primary host unwraps the key using the rsa key each time // access to the key is required, such as at server startup. // 2. the secondary host generates an rsa keypair that will be used // to transport keys between the primary host and itself. this // key needs to exist long enough to be used to process the // response to a key transport request that is made to the primary // server. the example here shows how to create a permanent (token) // rsa key for this purpose. (this key will also be used for // storage of the keys, since nss does not support permanent symmetric // keys...

Overview of NSS
eroperability and open standards you can use nss to support a range of security standards in your application, including the following: ssl v3. the secure sockets layer (ssl) protocol allows mutual authentication between a client and server and the establishment of an authenticated and encrypted connection. tls v1.3 (rfc 8446), tls v1.2 (rfc 5246), tls v1.1 (rfc 4346), tls v1 (rfc 2246). the transport layer security (tls) protocol from the ietf that supersedes ssl. pkcs #1. rsa standard that governs implementation of public-key cryptography based on the rsa algorithm. pkcs #3. rsa standard that governs implementation of diffie-hellman key agreement. pkcs #5. rsa standard that governs password-based cryptography, for example to encrypt private keys for storage. pkcs #7. rsa standard that go...
...ding those used with pkcs #7, pkcs #8, and pkcs #10. pkcs #10. rsa standard that governs the syntax for certificate requests. pkcs #11. rsa standard that governs communication with cryptographic tokens (such as hardware accelerators and smart cards) and permits application independence from specific algorithms and implementations. pkcs #12. rsa standard that governs the format used to store or transport private keys, certificates, and other secret material. s/mime (rfc 2311 and rfc 2633). ietf message specification (based on the popular internet mime standard) that provides a consistent way to send and receive signed and encrypted mime data. x.509 v3. itu standard that governs the format of certificates used for authentication in public-key cryptography. ocsp (rfc 2560). the online certificat...

sslfnc.html
ish a connection with a peer that only understands ssl v3 will fail. ssl_enable_ssl2 enables the application to communicate with ssl v2. factory setting is on. if you turn this option off, an attempt to establish a connection with a peer that only understands ssl v2 will fail. ssl_enable_tls is a peer of the ssl_enable_ssl2 and ssl_enable_ssl3 options. the ietf standard transport layer security (tls) protocol, rfc 2246, is a modified version of ssl3. it uses the ssl version number 3.1, appearing to be a "minor" revision of ssl 3.0. nss 2.8 supports tls in addition to ssl2 and ssl3. you can think of it as "ssl_enable_ssl3.1". see the description below for more information about this option. ssl_v2_compatible_hello tells the ssl library whether or not to send s...
... this option off, an attempt to establish a connection with a peer that understands only ssl v3 will fail. ssl_enable_ssl2 enables the application to communicate with ssl v2. if you turn this option off, an attempt to establish a connection with a peer that understands only ssl v2 will fail. ssl_enable_tls is a peer of the ssl_enable_ssl2 and ssl_enable_ssl3 options. the ietf standard transport layer security (tls) protocol, rfc 2246, is a modified version of ssl3. it uses the ssl version number 3.1, appearing to be a "minor" revision of ssl3.0. nss 2.8 supports tls in addition to ssl2 and ssl3. you can think of it as "ssl_enable_ssl3.1." see the description below for more information about this option. ssl_v2_compatible_hello tells the ssl library whether or not to send ssl3 clie...

Necko Architecture
ng the protocol handler to create the url. the nsiioservice can also create channels. it does this by looking up the protocol handler and asking it to create the channel. the primary responsibility of protocol handlers is to create channel instances that know how to interpret the protocol that the handler registered for. however, a protocol handler implementation also might "cache" the underlying transports so channels it creates can reuse them later. the channel does the true work of protocol interpretation and data movement (if any). although the protocol handler creates channels, protocol interpretation doesn't begin until the user initiates the transaction using the nsichannel api. receiving data & nsistreamlistener you can read or write, from or to a channel using either the synchronous api...
...you when things are happening. in the case of an asynchronous read... when you receive the ondataavailable() callback, you are handed an nsiinputsteam which you can read the data from (you will more than likely need to be able to handle multiple ondataavailable() callbacks, buffering the data you receive as necessary). the transaction is not complete until you receive an "stopped" notification. transports underlying i/o functionality has been consolidated into "transports." there is a transport for each type of i/o. necko implements a file transport and a socket transport that the protocol implementations (channels) use to actually move bytes to and from. how do i add my own protocol handler? adding your own protocol handler is a straightforward process in necko. there are a minimum of two in...

Proxies in Necko
eturned. if pac (proxy autoconfiguration) is enabled, it is asked for the proxy string for the given uri, which determines whether to use a proxy. note that the proxy method is only used when the protocol handler supports http proxies, as indicated by its protocol flags. finally, depending on the protocol, the proxy info will be created with the appropriate type, host and port. socks and nsisockettransportservice the aforementioned methods work very well for application-level proxies. however, socks is transparent to upper-level protocols, and can transport any other tcp- or udp-based protocol. therefore, nsisockettransportservice supports creating socket transports using an nsiproxyinfo. this proxy info will only be used if it specifies a socks proxy, through which the connection is then made. not...
...vel proxies. however, socks is transparent to upper-level protocols, and can transport any other tcp- or udp-based protocol. therefore, nsisockettransportservice supports creating socket transports using an nsiproxyinfo. this proxy info will only be used if it specifies a socks proxy, through which the connection is then made. note that socks is implemented as an nsisocketprovider, and the socket transport service will therefore use it as as the downmost socket type. proxies and local hosts requires gecko 9.0(firefox 9.0 / thunderbird 9.0 / seamonkey 2.6) starting with gecko 9.0 (firefox 9.0 / thunderbird 9.0 / seamonkey 2.6), proxy servers are not used when accessing hosts with no dots in their names, if manual proxy configuration is used and "<local>" is added to the "no proxy for:" tex...

nsIBadCertListener2
autf8string targetsite); methods notifycertproblem() called in case of a broken ssl 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. this object usually supports nsisslsocketcontrol, nsitransportsecurityinfo, nsiidentityinfo, nsisslstatusprovider, nsiclientauthuserdecision and possibly other interfaces. status the ssl status object that describes the problem(s). 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). see al...
... nsiidentityinfo, nsisslstatusprovider, nsiclientauthuserdecision and possibly other interfaces. status the ssl status object that describes the problem(s). 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). see also nsisockettransport ...

nsICacheEntryDescriptor
. return value returns the string value containing the meta data for the provided key. markvalid() this method means that a writer must validate this cache object before any readers are given a descriptor to the object. void markvalid(); parameters none. openinputstream() this method opens blocking input stream to cache data. this will fail if the cache entry is not stream based. use the stream transport service to asynchronously read this stream on a background thread. the returned stream may implement nsiseekablestream. nsiinputstream openinputstream( in unsigned long offset ); parameters offset read starting from this offset into the cached data. an offset beyond the end of the stream has undefined consequences. return value returns blocking, unbuffered input stream. openoutputstream() th...
...inputstream openinputstream( in unsigned long offset ); parameters offset read starting from this offset into the cached data. an offset beyond the end of the stream has undefined consequences. return value returns blocking, unbuffered input stream. openoutputstream() this method opens blocking output stream to cache data. this will fail if the cache entry is not stream based. use the stream transport service to asynchronously write to this stream on a background thread. the returned stream may implement nsiseekablestream. if opening an output stream to existing cached data, the data will be truncated to the specified offset. nsioutputstream openoutputstream( in unsigned long offset ); parameters offset write starting from this offset into the cached data. an offset beyond the end of the s...

nsIHttpChannelInternal
long major, out unsigned long minor ); parameters major on return, contains the response's major version number. minor on return, contains the response's minor version number. httpupgrade() allows for the use of http to bootstrap another protocol via the rfc 2616 upgrade request header in conjunction with a 101 level response. ...
...umber. minor on return, contains the response's minor version number. httpupgrade() allows for the use of http to bootstrap another protocol via the rfc 2616 upgrade request header in conjunction with a 101 level response. the nsihttpupgradelistener will have its nsihttpupgradelistener.ontransportavailable() method invoked if a matching 101 is processed. the arguments to nsihttpupgradelistener.ontransportavailable() provide to the new protocol the low level transport streams that are no longer used by http. the onstartrequest and onstoprequest events are still delivered and the listener gets full control over the socket if and when nsihttpupgradelistener.ontransportavailable() is delivered. void httpupgrade( in acstring aprotocolname, in nsihttpupgradelistener alistener ); parameters aprotoc...

nsIHttpUpgradeListener
interface for nsihttpchannelinternal.httpupgrade; this is called when an http protocol upgrade is finished. 1.0 66 introduced gecko 6.0 inherits from: nsisupports last changed in gecko 6.0 (firefox 6.0 / thunderbird 6.0 / seamonkey 2.3) this is used, for example, by websockets in order to upgrade an http channel to use the websocket protocol. method overview void ontransportavailable(in nsisockettransport atransport, in nsiasyncinputstream asocketin, in nsiasyncoutputstream asocketout); methods ontransportavailable() called when an http protocol upgrade attempt is completed, passing in the information needed by the protocol handler to take over the channel that is no longer being used by http. note: to begin a protocol upgrade, call nsihttpchannelinternal.htt...
...kettransport atransport, in nsiasyncinputstream asocketin, in nsiasyncoutputstream asocketout); methods ontransportavailable() called when an http protocol upgrade attempt is completed, passing in the information needed by the protocol handler to take over the channel that is no longer being used by http. note: to begin a protocol upgrade, call nsihttpchannelinternal.httpupgrade(). 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...

XPCOM Interface Reference
crollablensisearchenginensisearchsubmissionnsisecuritycheckedcomponentnsiseekablestreamnsiselectionnsiselection2nsiselection3nsiselectioncontrollernsiselectionimageservicensiselectionprivatensiserversocketnsiserversocketlistenernsiservicemanagernsisessionstartupnsisessionstorensisimpleenumeratornsismsdatabaseservicensismsrequestmanagernsismsservicensisocketprovidernsisocketproviderservicensisockettransportnsisockettransportservicensisoundnsispeculativeconnectnsistackframensistandardurlnsistreamconverternsistreamlistenernsistringbundlensistringbundleoverridensistringbundleservicensistringenumeratornsistructuredclonecontainernsistylesheetservicensisupportsnsisupports proxiesnsisupportsarraynsisupportscstringnsisupportscharnsisupportsdoublensisupportsfloatnsisupportsidnsisupportsinterfacepointernsisup...
...eviewnsitelemetrynsitextinputprocessornsitextinputprocessorcallbacknsitextinputprocessornotificationnsithreadnsithreadeventfilternsithreadinternalnsithreadmanagernsithreadobservernsithreadpoolnsithreadpoollistenernsitimernsitimercallbacknsitoolkitnsitoolkitprofilensitoolkitprofileservicensitraceablechannelnsitransactionnsitransactionlistnsitransactionlistenernsitransactionmanagernsitransferablensitransportnsitransporteventsinknsitransportsecurityinfonsitreeboxobjectnsitreecolumnnsitreecolumnsnsitreecontentviewnsitreeselectionnsitreeviewnsiurinsiurifixupnsiurifixupinfonsiurlnsiurlformatternsiurlparsernsiutf8converterservicensiutf8stringenumeratornsiuuidgeneratornsiupdatensiupdatechecklistenernsiupdatecheckernsiupdateitemnsiupdatemanagernsiupdatepatchnsiupdatepromptnsiupdatetimermanagernsiuploadchann...

XPCOM Interface Reference by grouping
erequestor nsijscid nsijsid nsijsiid nsimodule nsiobserver nsiobserverservice nsiproperties nsiproperty nsipropertybag nsipropertybag2 nsipropertyelement nsiserversocket nsiserversocketlistener nsiservicemanager nsisocketprovider nsisocketproviderservice nsisockettransport nsisockettransportservice nsisupports nsiuuidgenerator debug nsistackframe device display nsiscreen nsiscreenmanager geolocation nsigeolocationprovider nsigeolocationupdate orientation ...
... filter nsiparentalcontrolsservice nsipermission nsipermissionmanager nsisecuritycheckedcomponent ssl nsisslerrorlistener stream stream nsipipe nsitraceablechannel nsitransport nsitransporteventsink nsitransportsecurityinfo timer nsitimer nsitimercallback ui windows nsitaskbarpreview nsitaskbarpreviewbutton nsitaskbarpreviewcontroller nsitaskbarprogress nsitaskbartabpreview nsitaskbarwindowpreview ...

PerformanceResourceTiming.responseEnd - Web APIs
example in the following example, the value of the *start and *end properties of all "resource" type events are logged. ...
... example in the following example, the value of the *start and *end properties of all "resource" type events are logged. ...

PublicKeyCredentialRequestOptions - Web APIs
prompting a specific text on the authenticator, etc.). methods none. ...
...[/* bytes sent from the server */]), rpid: "example.com", /* will only work if the current domain is something like foo.example.com */ userverification: "preferred", timeout: 60000, // wait for a minute allowcredentials: [ { transports: "usb", type: "public-key", id: new uint8array(26) // actually provided by the server }, { transports: "internal", type: "public-key", id: new uint8array(26) // actually provided by the server } ], extensions: { uvm: true, // rp wants to know how the user was verified loc: false, txauthsimple: "could you please verify yourself?" } }; navigator.credentials.get({ "publickey": options }) .then(function (credentialinfoassertion) { // send assertion respons...

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. different browsers have different limitations on the size of the message you can send. ...
...plicated over time, but for now, if you have questions, see understanding message size limits in using webrtc data channels. syntax rtcdatachannel.send(data); parameters data the data to transmit across the connection. this may be a usvstring, a blob, an arraybuffer, or an arraybufferview. return value undefined. 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. networkerror the specified data would need to be buffered, and there isn't room for it in the buffer. in this scenario, the underlying transport is immediately closed. typeerror the specified da...

RTCIceCandidate.address - Web APIs
rn potentially reveal the address to the user. this can occur without the remote peer's consent. applications being built with user privacy and security in mind can choose to limit the permitted candidates to relay candidates only. doing so prevents the remote user's address from being exposed, but reduces the pool of available candidates to choose from. ...
...tes to relay candidates only. doing so prevents the remote user's address from being exposed, but reduces the pool of available candidates to choose from. to do this, configure the ice agent's ice transport policy using rtcconfiguration, like this: var rtcconfig = { iceservers: [ { urls: "turn:myturn.server.ip", username: "username", credential: "password" } ], icetransportpolicy: "relay" } by setting rtcconfiguration.icetransportpolicy to "relay", any host candidates (candidates where the ip address is the peer's own ip address) are left out of the pool of candidates, as are any other candidates which aren't relay candidates. usage notes consider this sdp attribute line (a-line) which describes an ice candidate: a=candidate:4234997325 1 udp 2043278322 192.168.

RTCIceCandidate - Web APIs
ured based on an object based on the rtcicecandidateinit dictionary. note: for backward compatibility, the constructor also accepts as input a string containing the value of the candidate property instead of a rtcicecandidateinit object, since the candidate includes all of the information that rtcicecandidateinit does and more. properties candidate read only a domstring representing the transport address for the candidate that can be used for connectivity checks. the format of this address is a candidate-attribute as defined in rfc 5245. this string is empty ("") if the rtcicecandidate is an "end of candidates" indicator. component read only a domstring which indicates whether the candidate is an rtp or an rtcp candidate; its value is either "rtp" or "rtcp", and is derived from the "c...
... rtcicecomponent enumerated type. foundation read only returns a domstring containing a unique identifier that is the same for any candidates of the same type, share the same base (the address from which the ice agent sent the candidate), and come from the same stun server. this is used to help optimize ice performance while prioritizing and correlating candidates that appear on multiple rtcicetransport objects. ip read only a domstring containing the ip address of the candidate. port read only an integer value indicating the candidate's port number. priority read only a long integer value indicating the candidate's priority. protocol read only a string indicating whether the candidate's protocol is "tcp" or "udp". the string is one of those in the enumerated type rtciceprotocol. re...

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. syntax localcandidate = rtcicecandidatepair.local; value an rtcicecandidate which describes the configuration of the local end of a viable pair of ice candidates. the rtcicecandidatepair is returned by the rtcicetransport method getselectedcandidatepair(). example this one-line example simply obtains the current candidate pair and then from that gets the local candidate. 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. in the first rtcrtpsende...
.... example this one-line example simply obtains the current candidate pair and then from that gets the local candidate. 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. in the first rtcrtpsender, we get the rtcdtlstransport over which the media data is being transmitted and finally, from that, the rtcicetransport. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecandidatepair.local' in that specification. candidate recommendation initial specification. ...

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. syntax remotecandidate = rtcicecandidatepair.remote; value an rtcicecandidate which describes the configuration of the remote end of a viable pair of ice candidates. the rtcicecandidatepair is returned by the rtcicetransport method getselectedcandidatepair(). example this one-line example simply obtains the current candidate pair and then from that gets the remote candidate. 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. in the first rtcrtpse...
...example this one-line example simply obtains the current candidate pair and then from that gets the remote candidate. 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. in the first rtcrtpsender, we get the rtcdtlstransport over which the media data is being transmitted and finally, from that, the rtcicetransport. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecandidatepair.remote' in that specification. candidate recommendation initial specification. ...

RTCIceComponent - Web APIs
the webrtc api's rtcicecomponent enumerated type contains domstring values that each identify a specific ice component; these are "rtp" and "rtcp". these strings are mapped to corresponding numeric values as they appear in the candidate a-line in sdp. values "rtp" identifies an ice transport which is being used for the real-time transport protocol (rtp), or for rtp multiplexed with the rtp control protocol (rtcp). rtp is defined in rfc 3550. an rtcicecomponent of this value corresponds to the component id field in the candidate a-line with the value 1. "rtcp" identifies an ice transport being used for rtcp, which is defined in rfc 3550, section 6. this value of rtcicecomponent corresponds to the component id 2. usage notes the rtcicecomponent type is used by the component property of rtc...
...efined in rfc 3550. an rtcicecomponent of this value corresponds to the component id field in the candidate a-line with the value 1. "rtcp" identifies an ice transport being used for rtcp, which is defined in rfc 3550, section 6. this value of rtcicecomponent corresponds to the component id 2. usage notes the rtcicecomponent type is used by the component property of rtcicecandidate and rtcicetransport objects. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicecomponent' in that specification. candidate recommendation initial definition. ...

RTCPeerConnection: icecandidate event - Web APIs
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. indicating the end of a generation of candidates when an ice negotiation session runs out of candidates to propose for a given rtcicetransport, it has completed gathering for a generation of candidates. that this has occurred is indicated by an icecandidate event whose candidate string is empty (""). you should deliver this to the remote peer just like any standard candidate, as described under sharing a new candidate above. this ensures that the remote peer is given the end-of-candidates notification as well. as you see in the code in...
...mpty candidate string. only candidates for which the event's candidate property is null are not forwarded across the signaling connection. the end-of-candidates indication is described in section 9.3 of the trickle ice draft specification (note that the section number is subject to change as the specification goes through repeated drafts). indicating that ice gathering is complete once all ice transports have finished gathering candidates and the value of the rtcpeerconnection object's icegatheringstate has made the transition to complete, an icecandidate event is sent with the value of complete set to null. this signal exists for backward compatibility purposes and does not need to be delivered onward to the remote peer (which is why the code snippet above checks to see if event.candidate is n...

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. syntax rtcpeerconnection.onconnectionstatechange = eventhandler; value a function which is called by the browser when the connectionstatechange event occurs on the rtcpeerconnection. the function receives as input a single parameter, which is an object of type event. the event object contains no special information of note; you can look at the value of the peer connection's connectionstate property to determine wha...
... object contains no special information of note; you can look at the value of the peer connection's connectionstate property to determine what the new state is. example pc.onconnectionstatechange = function(event) { switch(pc.connectionstate) { case "connected": // the connection has become fully connected break; case "disconnected": case "failed": // one or more transports has terminated unexpectedly or in an error break; case "closed": // the connection has been closed break; } } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.onconnectionstatechange' in that specification. candidate recommendation initial specification.

WebRTC connectivity - Web APIs
ddress is local to the remote peer. the stun server then replies to the initiator's request with a candidate whose ip address is unrelated to the remote peer. relay a relay candidate is generated just like a server reflexive candidate ("srflx"), but using turn instead of stun. tcp candidate types tcp candidates (that is, candidates whose protocol is tcp) can be of these types: active the transport will try to open an outbound connection but won't receive incoming connection requests. this is the most common type, and the only one that most user agents will gather. passive the transport will receive incoming connection attempts but won't attempt a connection itself. so the transport will try to simultaneously open a connection with its peer. choosing a candidate pair the ice layer se...
...ly open a connection with its peer. choosing a candidate pair the ice layer selects one of the two peers to serve as the controlling agent. this is the ice agent which will make the final decision as to which candidate pair to use for the connection. the other peer is called the controlled agent. you can identify which one your end of the connection is by examining the value of rtcicecandidate.transport.role, although in general it doesn't matter which is which. the controlling agent not only takes responsibility for making the final decision as to which candidate pair to use, but also for signaling that selection to the controlled agent by using stun and an updated offer, if necessary. the controlled agent just waits to be told which candidate pair to use. it's important to keep in mind that ...

Web APIs
opstateevent positionoptions processinginstruction progressevent promiserejectionevent publickeycredential publickeycredentialcreationoptions publickeycredentialrequestoptions pushevent pushmanager pushmessagedata pushregistrationmanager pushsubscription r rtcansweroptions rtccertificate rtcconfiguration rtcdtmfsender rtcdtmftonechangeevent rtcdatachannel rtcdatachannelevent rtcdtlstransport rtcerror rtcerrorevent rtcicecandidate rtcicecandidateinit rtcicecandidatepair rtcicecandidatepairstats rtcicecandidatestats rtcicecandidatetype rtcicecomponent rtcicecredentialtype rtcicegathererstate rtciceparameters rtciceprotocol rtcicerole rtciceserver rtcicetcpcandidatetype rtcicetransport rtcicetransportstate rtcidentityassertion rtcidentityerrorevent rtcidentityevent rtcinboundr...
...tcpeerconnectioniceevent rtcremoteoutboundrtpstreamstats rtcrtcpparameters rtcrtpcapabilities rtcrtpcodeccapability rtcrtpcodecparameters rtcrtpcontributingsource rtcrtpencodingparameters rtcrtpparameters rtcrtpreceiveparameters rtcrtpreceiver rtcrtpsendparameters rtcrtpsender rtcrtpstreamstats rtcrtpsynchronizationsource rtcrtptransceiver rtcrtptransceiverdirection rtcrtptransceiverinit rtcsctptransport rtcsessiondescription rtcsessiondescriptioncallback rtcstats rtcstatsicecandidatepairstate rtcstatsreport rtcstatstype rtctrackevent rtctrackeventinit radionodelist range readablebytestreamcontroller readablestream readablestreambyobreader readablestreambyobrequest readablestreamdefaultcontroller readablestreamdefaultreader relativeorientationsensor renderingcontext report reporting...

Evolution of HTTP - HTTP
e company, but ssl 2.0 and its successor ssl 3.0 allowed for the creation of e-commerce web sites by encrypting and guaranteeing the authenticity of the messages exchanged between the server and client. ssl was put on the standards track and eventually became tls, with versions 1.0, 1.1, 1.2, and 1.3 appearing successfully to close vulnerabilities. during the same time, the need for an encrypted transport layer raised: the web left the relative trustiness of a mostly academic network, to a jungle where advertisers, random individuals or criminals compete to get as much private information about people, try to impersonate them or even to replace data transmitted by altered ones. as the applications built over http became more and more powerful, having access to more and more private information lik...
... see bug 1158011 for implementation status in firefox. ...

HTTP Index - HTTP
etween web browsers and web servers, but it can also be used for other purposes. http follows a classical client-server model, with a client opening a connection to make a request, then waiting until it receives a response. http is a stateless protocol, meaning that the server does not keep any data (state) between two requests. though often based on a tcp/ip layer, it can be used on any reliable transport layer, that is, a protocol that doesn't lose messages silently like udp does. rudp — the reliable update of udp — is a suitable alternative. 2 a typical http session http in client-server protocols, like http, sessions consist of three phases: 3 an overview of http html, http, overview, webmechanics, l10n:priority http isthe foundation of any data exc...
...rom the server to the user agent, but has been deprecated by the specification. use set-cookie instead. 183 sourcemap http, http header, reference, response header, header the sourcemap http response header links generated code to a source map, enabling the browser to reconstruct the original source and present the reconstructed original in the debugger. 184 strict-transport-security hsts, http, https, security, header the http strict-transport-security response header (often abbreviated as hsts) lets a web site tell browsers that it should only be accessed using https, instead of using http. 185 te http, reference, header the te request header specifies the transfer encodings the user agent is willing to accept. (you could informally c...

Media container formats (file types) - Web media technologies
ntainer format, find it in this list and click through to the details, which include information about what the container is typically useful for, what codecs it supports, and which browsers support it, among other specifics. ...
... yes1,2 he-aac v1 yes1,2 he-aac v2 yes1,2 mp3 yes1 [1] firefox only supports 3gp on openmax-based devices, which currently means the boot to gecko (b2g) platform. [2] firefox support for aac relies upon the operating system's media infrastructure, so it is available as long as the os supports it. adts audio data transport stream (adts) is a container format specified by mpeg-4 part 3 for audio data, intended to be used for streamed audio, such as for internet radio. it is, essentially, an almost bare stream of aac audio data, comprised of adts frames with a minimal header. adts media mime types audio audio/aac[1] audio/mpeg[1] [1] the mime type used for adts depends on what ...

Privacy, permissions, and information security
ute technically part of feature policy, the allow attribute on an <iframe> specifies which web features the document in the frame should be allowed to access http public key pinning (hpkp) hpkp is used by servers to instruct a client to associate a specific public key with the server going forward in order to decrease the likelihood of man-in-the-middle attacks http strict transport security (hsts) hsts is used by servers to let them protect themselves from protocol downgrade and cookie hijack attacks by letting sites tell clients that they can only use https to communicate with the server http/2 while http/2 technically does not have to use encryption, most browser developers are only supporting it when used with https, so it can be thought of in that regard ...
...ients that they can only use https to communicate with the server http/2 while http/2 technically does not have to use encryption, most browser developers are only supporting it when used with https, so it can be thought of in that regard as being security-related permissions api provides a way to determine the status of permissions for the current browser context transport layer security (tls); formerly known as secure sockets layer (ssl) tls provides security and privacy by encrypting data during transport over the network. this is the technology behind the https (hypertext transport protocol secured) protocol not all of these are generally directly used within code; notably, the permissions api, feature policy, and the allow attribute on <iframe> elemen...

Web security
de from attacks and data theft. content security content security policy (csp) content security policy (csp) is an added layer of security that helps to detect and mitigate certain types of attacks, including cross-site scripting (xss) and data injection attacks. these attacks are used for everything from data theft to site defacement to the distribution of malware. 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. applications that use tls can choose their security parameters, which can have a substantial impact on the security and reliability of data. this article provides an overview of tls and the kinds of decisions you need ...
... of tls and the kinds of decisions you need to make when securing your content. https https (hypertext transfer protocol secure) is an encrypted version of the http protocol. it uses ssl or tls to encrypt all communication between a client and a server. this secure connection allows clients to be sure that they are connected with the intended server, and to exchange sensitive data. http strict-transport-security the strict-transport-security: http header lets a website specify that it may only be accessed using https. certificate transparency certificate transparency is an open framework designed to protect against and monitor for certificate misissuances. newly issued certificates are 'logged' to publicly run, often independent ct logs which maintain an append-only, cryptographically assured...

Index - Archive of obsolete content
l has been universally accepted on the world wide web for authenticated and encrypted communication between clients and servers. 3745 nspr release engineering guide nspr this paper is for engineers performing formal release for the netscape portable runtime (nspr) across all platforms. 3746 ssl and tls security, tutorial the secure sockets layer (ssl) and transport layer security (tls) protocols are universally accepted standards for authenticated and encrypted communication between clients and servers. both client and server authentication occur over ssl/tls. 3747 security controls beginner, security, tutorial sensitive data should be protected based on the potential impact of a loss of confidentiality, integrity, or availability. prot...

Inner-browsing extending the browser navigation paradigm - Archive of obsolete content
rowsing and possible approaches for its implementation. it also provides two implementation examples using the model. the first one looks like a traditional dhtml ticker application and uses hidden iframes and javascript to provide updates inside the webpage's ticker headlines section. this sample was made using iframes to illustrate a possible implementation. an implementation can also use other transport technologies such as soap-based remote procedure calls, xmlhttprequest interfaces, or dom 3 load. the second example shows a tabbed menu that dynamically loads news headlines into the current web page. this implementation uses xmlhttprequest. extending traditional hypertext navigation the hypertext approach to developing web pages, in which developers format text layouts into pages and link those...

HTTP Class Overview - Archive of obsolete content
. ...

JavaScript crypto - Archive of obsolete content
<form name="reqform" method="post" action="http://your.ra.site.org"> <p><input type=hidden name=cert_request value=""> <p>name: <input type=text name=name value=""> <p>password: <input type=password name="password" value=""> <p><input type=submit name="send" value="submit"> </form> <script> /* the following values could be filled in by the server cgi */ var authenticator = "server_magic"; var keytransportcert = null; var crmfobject = null; var form = document.forms[0]; function validate() { // generate keys for nsm. if (typeof(crypto.version) != "undefined") { crmfobject = crypto.generatecrmfrequest( "cn=" + form.name.value, form.password.value, authenticator, keytransportcert, "setcrmfrequest();", 1024, null, "rsa-dual-use"); } return f...

New Security Model for Web Services - Archive of obsolete content
ers from scripts loaded from the domain with host name containing mozilla.org. that is, http://www.mozilla.org/, http://lxr.mozilla.org, http://komodo.mozilla.org, etc. will be granted access. implementation nsiwebscriptsaccessservice this interface provides a way to check whether the running script has access to the server that the script wishes to communicate. boolean canaccess(in nsiuri atransporturi, in astring atype); atransporturi - the service uri atype - type requested by the script ( ex. soapv, soap, load, etc. ) return pr_true if access granted else pr_false void invalidatecache(in string atransporturi); invalidate cached entry for the given transport uri. empty the cache by passing in a empty string nswebscriptsaccess(implements nsiwebscriptsaccess...

SSL and TLS - Archive of obsolete content
both client and server authentication occur over ssl/tls. ssl/tls uses a combination of public key and symmetric-key encryption. symmetric-key encryption is much faster than public-key encryption, but public-key encryption provides better authentication techniques. an ssl/tls session always begins with an exchange of messages called the ssl handshake, initial communication between the server and client. ...

Security - Archive of obsolete content
ction to sslthis document introduces the secure sockets layer (ssl) protocol. ssl has been universally accepted on the world wide web for authenticated and encrypted communication between clients and servers.nspr release engineering guidethis paper is for engineers performing formal release for the netscape portable runtime (nspr) across all platforms.ssl and tlsthe secure sockets layer (ssl) and transport layer security (tls) protocols are universally accepted standards for authenticated and encrypted communication between clients and servers. both client and server authentication occur over ssl/tls. ...

Base64 - MDN Web Docs Glossary: Definitions of Web-related terms
in an ascii string format by translating it into a radix-64 representation. the term base64 originates from a specific mime content transfer encoding. base64 encoding schemes are commonly used when there is a need to encode binary data that needs to be stored and transferred over media that are designed to deal with ascii. this is to ensure that the data remain intact without modification during transport. base64 is commonly used in a number of applications including email via mime, and storing complex data in xml. one common application of base64 encoding on the web is to encode binary data so it can be included in a data: url. in javascript there are two functions respectively for decoding and encoding base64 strings: btoa(): creates a base-64 encoded ascii string from a "string" of binary...

Domain sharding - MDN Web Docs Glossary: Definitions of Web-related terms
ed to be downloaded sequentially. ...

HTTP - MDN Web Docs Glossary: Definitions of Web-related terms
the hypertext transfer protocol (http) is the underlying network protocol that enables transfer of hypermedia documents on the web, typically between a browser and a server so that humans can read them. the current version of the http specification is called http/2. as part of a uri, the "http" within "http://example.com/" is called a "scheme". resources using the "http" schema are typically transported over unencrypted connections using the http protocol. the "https" scheme (as in "https://developer.mozilla.org") indicates that a resource is transported using the http protocol, but over a secure tls channel. http is textual (all communication is done in plain text) and stateless (no communication is aware of previous communications). this property makes it ideal for humans to read documents ...

HTTP/2 - MDN Web Docs Glossary: Definitions of Web-related terms
tiplexing, minimize protocol overhead via efficient compression of http header fields, and add support for request prioritization and server push. http/2 does not modify the application semantics of http in any way. all the core concepts found in http 1.1, such as http methods, status codes, uris, and header fields, remain in place. instead, http/2 modifies how the data is formatted (framed) and transported between the client and server, both of which manage the entire process, and hides application complexity within the new framing layer. as a result, all existing applications can be delivered without modification. ...

QUIC - MDN Web Docs Glossary: Definitions of Web-related terms
it was developed by google as a way to experiment with ways to improve tcp and web application delivery. as tcp is built into the kernel of many operating systems being able to experiment with changes, test them and implement modifications is an extremely slow process. the creation of quic allows developers to conduct experiments and try new things faster. quic was designed to support the semantics of http/2. it provides multiplexing, flow control, security and congestion control. key features of quic include: reduction in connection establishment time. improved congestion control. ...

RTCP (RTP Control Protocol) - MDN Web Docs Glossary: Definitions of Web-related terms
ll as the rtp and media contents. ...

SCTP - MDN Web Docs Glossary: Definitions of Web-related terms
it's used for sending traditional telephone calls over the internet, but is also used for webrtc data. ...

Secure Sockets Layer (SSL) - MDN Web Docs Glossary: Definitions of Web-related terms
...

TCP handshake - MDN Web Docs Glossary: Definitions of Web-related terms
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 messages transmitted by tcp to negotiate and start a tcp session are nicknamed syn, syn-ack, and ack for synchronize, synchronize-acknowledgement, and acknowledge respectively. 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. ...

TOFU - MDN Web Docs Glossary: Definitions of Web-related terms
for identifiers (for example public keys) stored locally. if an identifier is found, the client can establish the connection. if no identifier is found, the client can prompt the user to determine if the client should trust the identifier. tofu is used in the ssh protocol, in http public key pinning (hpkp) where the browsers will accept the first public key returned by the server, and in strict-transport-security (hsts) where a browser will obey the redirection rule. learn more http public key pinning (hpkp) public-key-pins wikipedia: tofu ...

Transmission Control Protocol (TCP) - MDN Web Docs Glossary: Definitions of Web-related terms
tcp uses virtual ports to create a virtual end-to-end connection that can reuse the physical connections between two computers. tcp encapsulates higher level protocol data such as http and, smtp (email) . 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 messages transmitted by tcp to negotiate and start a tcp session are nicknamed syn, syn-ack, ack for synchronize, synchronize-acknowledgement, and acknowledge. ...

How the Web works - Learn web development
o buy something from. 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. it's basically like the street between your house and the shop. tcp/ip: transmission control protocol and internet protocol are communication protocols that define how data should travel across the internet. this is like the transport mechanisms that let you place an order, go to the shop, and buy your goods. in our example, this is like a car or a bike (or however else you might get around). dns: domain name servers are like an address book for websites. when you type a web address in your browser, the browser looks at the dns to find the website's real address before it can retrieve the website. the browser needs to find ou...

The "why" of web performance - Learn web development
e loading experience of cnn.com, which at the time of this writing had over 400 http requests with a file size of over 22.6mb. imagine loading this on a desktop computer connected to a fibre optic network. this would seem relatively fast, and the file size would be largely irrelevant. imagine loading that same site using tethered mobile data on a 9 year-old ipad while commuting home on public transportation. the same site will be slow to load, possibly verging on unusable depending on cell coverage. you might give up before it finishes loading. imagine loading that same site on a $35 huawei device in a rural india with limited coverage or no coverage. the site will be very slow to load—if it loads at all—with blocking scripts possibly timing out, and adverse cpu impact causing browser cras...

Website security - Learn web development
password management. encourage strong passwords that are changed regularly. consider two-factor authentication for your site, so that in addition to a password the user must enter another authentication code (usually one that is delivered via some physical hardware that only the user will have, such as a code in an sms sent to their phone). configure your web server to use https and http strict transport security (hsts). https encrypts data sent between your client and server. this ensures that login credentials, cookies, post requests data and header information are not easily available to attackers. keep track of the most popular threats (the current owasp list is here) and address the most common vulnerabilities first. use vulnerability scanning tools to perform automated security testing on...

Listening to events on all tabs
fired the notification. awebprogress the nsiwebprogress instance that fired the notification. arequest the nsirequest that has new status. astatus this value is not an error code. instead, it is a numeric value that indicates the current status of the request. this interface does not define the set of possible status codes. note: some status values are defined by nsitransport and nsisockettransport. amessage localized text corresponding to astatus. onrefreshattempted notification that a refresh or redirect has been requested in awebprogress for example, via a <meta http-equiv="refresh"> or an http refresh: header. if any registered progress listener returns false from this method then the attempt to refresh will be blocked. boolean onrefreshattempted( nsi...

IPDL Tutorial
see ipdl processes and threads for more details. because protocol messages are represented as c++ methods, it's easy to forget that they are in fact asynchronous messages: by default the c++ method will return immediately, before the message has been delivered. the parameters of the recv* methods (const nscstring& pluginpath in the example) are refer...

Examples
function promisevalueaftertimeout(avalue, atimeout) { let deferred = promise.defer(); try {...

NSS Sample Code sample2
nss sample code 2: symmetric encryption /* example code to illustrate des enccryption/decryption using nss. * the example skips the details of obtaining the key & iv to use, and * just uses a hardcoded key & iv. * note: iv is only needed if cipher blocking chaining (cbc) mode of encryption * is used * * the recommended approach is to store and transport wrapped (encrypted) * des keys (ivs can be in the clear). however, it is a common (and dangerous) * practice to use raw des keys. this example shows the use of a raw key. */ #include "nss.h" #include "pk11pub.h" /* example key & iv */ unsigned char gkey[] = {0xe8, 0xa7, 0x7c, 0xe2, 0x05, 0x63, 0x6a, 0x31}; unsigned char giv[] = {0xe4, 0xbb, 0x3b, 0xd3, 0xc3, 0x71, 0x2e, 0x58}; int main(int argc, char **argv) { ck_mechan...

NSS Sample Code
nss sample code the collection of sample code here demonstrates how nss can be used for cryptographic operations, certificate handling, ssl, etc. it also demonstrates some best practices in the application of cryptography. old samples in the process of being replaced. sample code 1: key generation and transport between servers sample code 2: symmetric encryption sample code 3: hashing, mac sample code 4: pki encryption sample code 5: pki encryption with a raw public & private key in der format sample code 6: persistent symmetric keys in nss database these are very old examples in need of replacement. see https://bugzilla.mozilla.org/show_bug.cgi?id=490238 you are welcome to download the new samples via: hg clone https://hg.mozilla.org/projects/nss; cd nss; hg update samples_...

sslerr.html
tempting to import certificates. sec_error_pkcs12_decoding_pfx -8114 unable to import. decoding error. file not valid. sec_error_pkcs12_invalid_mac -8113 unable to import. invalid mac. incorrect password or corrupt file. sec_error_pkcs12_unsupported_mac_algorithm -8112 unable to import. mac algorithm not supported. sec_error_pkcs12_unsupported_transport_mode -8111 unable to import. only password integrity and privacy modes supported. sec_error_pkcs12_corrupt_pfx_structure -8110 unable to import. file structure is corrupt. sec_error_pkcs12_unsupported_pbe_algorithm -8109 unable to import. encryption algorithm not supported. sec_error_pkcs12_unsupported_version -8108 unable to import. file versio...

Necko Interfaces Overview
nsirequest attribute of a nsirequest 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 las...

Necko
this api is used in many mozilla-based client applications (including firefox) and can be used for writing other networking clients. currently the library is built as part of the mozilla distribution. you can get the whole distribution as described in the build documentation. eventually we'd like to move to a binary distribution mechanism for the networking library so that you can build your application that uses this library without having to build mozilla. ...

Mozilla Projects
ng, leaving the reader to wonder....is this specific check necessary for (e.g.) an operating system? because information is not consolidated, checks are not done consistently, nor is it defined what we are checking for. necko necko is a network library that provides a platform-independent api for several layers of networking, ranging from transport to presentation layers. this api is used in many mozilla-based client applications (including firefox) and can be used for writing other networking clients. network security services network security services (nss) is a set of libraries designed to support cross-platform development of security-enabled client and server applications. appl...

Monitoring HTTP activity
components.interfaces.nsihttpactivitydistributor); activitydistributor.addobserver(httpobserver); observable activities there are two classes of observable activities: those that occur at the socket level and those that occur at the http transaction level. observable socket activities when the activity type reported to your nsihttpactivityobserver.observeactivity() method is activity_type_socket_transport, the activity subtype, which indicates the specific type of activity that occurred, will be a socket transport status code. observable http activities when the activity type is activity_type_http_transaction, the activity subtype will be one of the activity subtype constants. these include the ability to monitor outgoing http request headers and bodies as well as incoming http headers and complet...

nsIChannel
he owner, corresponding to the entity that is responsible for this channel. used by the security manager to grant or deny privileges to mobile code loaded from this channel. note: this is a strong reference to the owner, so if the owner is also holding a strong reference to the channel, care must be taken to explicitly drop its reference to the channel. securityinfo nsisupports transport-level security information (if any, else null) corresponding to the channel, normally presented through the interfaces nsitransportsecurityinfo and nsisslstatusprovider read only. uri nsiuri the uri corresponding to the channel. its value is immutable. read only. constants channel specific load flags: bits 23-31 are reserved for future use by this interface or one of its d...

nsIProgressEventSink
n nsresult astatus, in wstring astatusarg ); parameters arequest the request being observed (may qi to nsichannel). acontext if arequest is a channel, then this parameter is the listener context passed to nsichannel.asyncopen(). astatus status code (not necessarily an error code) indicating the state of the channel (usually the state of the underlying transport). see nsisockettransport for socket specific status codes. astatusarg status code argument to be used with the string bundle service to convert the status message into localized, human readable text. the meaning of this parameter is specific to the value of the status code. for socket status codes, this parameter indicates the host:port associated with the status code. ...

nsIProtocolProxyService
n nsicancelableobject that can be used to cancel the asynchronous operation. if canceled, the cancellation status (areason) will be forwarded to the callback's nsiprotocolproxycallback.onproxyavailable() method via the astatus parameter. newproxyinfo() this method may be called to construct a nsiproxyinfo instance from the given parameters. this method may be useful in conjunction with nsisockettransportservice.createtransport() for creating, for example, a socks connection. nsiproxyinfo newproxyinfo( in acstring atype, in autf8string ahost, in long aport, in unsigned long aflags, in unsigned long afailovertimeout, in nsiproxyinfo afailoverproxy ); parameters atype the proxy type is a string value that identifies the proxy type. standard values are: "http" - specifies a htt...

nsIRequest
annel implementations are only pending once asyncopen returns successfully). requests can become pending multiple times during their lifetime. note: suspended requests are still considered pending. boolean ispending(); parameters none. return value true if the request has yet to reach completion. resume() resumes the current request. this may have the effect of re-opening any underlying transport and will resume the delivery of data to any open streams. void resume(); parameters none. suspend() suspends the current request. this may have the effect of closing any underlying transport (in order to free up resources), although any open streams remain logically opened and will continue delivering data when the transport is resumed. calling cancel() on a suspended request must not send...

nsISSLErrorListener
be used to obtain more information about the active connection. error the code associated with the error. the ssl status object that describes the problem(s). 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). see also nsisockettransport ...

nsIServerSocket
socket in the listening state. it will asynchronously listen for and accept client connections. the listener will be notified once for each client connection that is accepted. the listener's onsocketaccepted() method will be called on the same thread that called asynclisten() (the calling thread must have an nsieventtarget). the listener will be passed a reference to an already connected socket transport (nsisockettransport). see nsiserversocketlistener for more details. void asynclisten( in nsiserversocketlistener alistener ); parameters alistener the listener to be notified when client connections are accepted. close() this method closes a server socket. this does not affect already connected client sockets (i.e., the nsisockettransport instances created from this server socket). th...

nsIWebProgressListener
atus, in wstring amessage ); parameters awebprogress the nsiwebprogress instance that fired the notification. arequest the nsirequest that has new status. astatus this value is not an error code. instead, it is a numeric value that indicates the current status of the request. this interface does not define the set of possible status codes. note: some status values are defined by nsitransport and nsisockettransport. amessage localized text corresponding to astatus. example the nsiwebprogresslistener for each tab: gbrowser.mtabprogresslistener can be used in the parent to listen for most nsiwebprogresslistener events, but in browser code, will not have access to the nsiwebprogress's domwindow property. see the example below on how to use an nsiwebprogresslistener in a frame scr...

nsIWebSocketChannel
originaluri nsiuri the original uri used to construct the protocol connection. this is used in the case of a redirect or uri "resolution" (for example resolving a resource: uri to a file: uri) so that the original pre-redirect uri can still be obtained. this is never null. read only. protocol acstring sec-websocket-protocol value. securityinfo nsisupports transport-level security information (if any). read only. uri nsiuri the uri corresponding to the protocol connection after any redirections are completed. read only. status codes the following values are permitted status codes. status code name description 0-999 reserved and not used. 1000 close_normal normal closure; the connection succe...

MailNews fakeserver
ned for use in automated tests. it can also be used in manual qa tests. basic structure fakeserver is situated entirely under mailnews/test/fakeserver/. it is comprised of three major components: a daemon, a handler, and the server itself. the server itself is found in mailnews/test/fakeserver/maild.js and was largely based off of the httpd fakeserver from network code. it utilizes utf-8 as its transport mechanism and is capable of performing proper pipelining of commands. the other two components are specific to the protocol type and found under similar files in the same directory (e.g., smtpd.js). a daemon is information about the server that can be manipulated. an example of such a manipulation is adding a folder or adding messages. daemons should usually persist for the entirety of the test. ...

Network request details - Firefox Developer Tools
"value": "true" }, { "name": "p3p", "value": "cp=\"this is not a p3p policy! ...

Console messages - Firefox Developer Tools
is a security risk that allows user login credentials to be stolen. forms containing password fields must submit them over https, not http. password fields present on an insecure (http://) iframe. this is a security risk that allows user login credentials to be stolen. iframes containing login forms must be served over https, not http. the site specified an invalid strict-transport-security header. see http strict transport security for more details. this site makes use of a sha-1 certificate; it's recommended you use certificates with signature algorithms that use hash functions stronger than sha-1. the site uses a certificate whose signature uses the sha-1 hash algorithm. sha-1 is still still widely used in certificates, but it is startin...

AudioListener.dopplerFactor - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.forwardX - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.forwardY - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.forwardZ - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.positionX - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.positionY - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.positionZ - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.setOrientation() - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.setPosition() - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.speedOfSound - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.upX - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.upY - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener.upZ - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AudioListener - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

AuthenticatorAttestationResponse - Web APIs
son secure contextread only client data for the authentication, such as origin and challenge. the clientdatajson property is inherited from the authenticatorresponse. authenticatorattestationresponse.attestationobject secure contextread only an arraybuffer containing authenticator data and an attestation statement for a newly-created key pair. methods authenticatorattestationresponse.gettransports()secure context returns an array of strings describing which transport methods (e.g. usb, nfc) are believed to be supported with the authenticator. the array may be empty if the information is not available. examples var publickey = { challenge: /* from the server */, rp: { name: "example corp", id : "login.example.com" }, user: { id: new uint8array(16), name: "jdoe@...

BaseAudioContext.createPanner() - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

CloseEvent - Web APIs
w closeevent. properties this interface also inherits properties from its parent, event. closeevent.code read only returns an unsigned short containing the close code sent by the server. the following values are permitted status codes. the following definitions are sourced from the iana website [ref]. note that the 1xxx codes are only websocket-internal and not for the same meaning by the transported data (like when the application-layer protocol is invalid). the only permitted codes to be specified in firefox are 1000 and 3000 to 4999 [source, bug]. status code name description 0–999 reserved and not used. 1000 normal closure normal closure; the connection successfully completed whatever purpose for which it was created. ...

PannerNode.distanceModel - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PannerNode.maxDistance - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PannerNode.panningModel - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PannerNode.setOrientation() - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PannerNode.setPosition() - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PannerNode.setVelocity() - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PannerNode - Web APIs
udio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the functions moveright(), moveleft(), etc., which set new values for the panner position via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those ...

PerformanceResourceTiming.connectEnd - Web APIs
note: this feature is available in web workers. syntax resource.connectend; return value a domhighrestimestamp representing the time after a connection is established. ...

PerformanceResourceTiming.requestStart - Web APIs
there is no end property for requeststart. note: this feature is available in web workers. ...

PerformanceResourceTiming - Web APIs
tely before the browser starts requesting the resource from the server. performanceresourcetiming.responsestartread only a domhighrestimestamp immediately after the browser receives the first byte of the response from the server. 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. performanceresourcetiming.transfersizeread only a number representing the size (in octets) of the fetched resource. the size includes the response header fields plus the response payload body. performanceresourcetiming.encodedbodysizeread only a number representing the size (in octets) received from the fetch (http or cache), of the payload body,...

PerformanceTiming.connectEnd - Web APIs
this interface of this property is deprecated in the navigation timing level 2 specification. please use the performancenavigationtiming interface instead. 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. syntax time = performancetiming.connectend; specifications specification status comme...

PerformanceTiming.connectStart - Web APIs
this interface of this property is deprecated in the navigation timing level 2 specification. please use the performancenavigationtiming interface instead. 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. syntax time = performancetiming.connectstart; specifications specification status comment navigation timingthe definition of 'performancetiming.connectstart' in tha...

PerformanceTiming.requestStart - Web APIs
f this property is deprecated in the navigation timing level 2 specification. please use the performancenavigationtiming interface instead. the legacy performancetiming.requeststart read-only property returns an unsigned long long representing the moment, in miliseconds since the unix epoch, when the browser sent the request to obtain the actual document, from the server or from a cache. 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. syntax time = performancetiming.requeststart; specifications specification status comment navigation timingthe definition of 'performancetiming.requeststart' in that specification. recommendation initial definition. ...

PublicKeyCredentialCreationOptions.attestation - Web APIs
the attestation is a mean for the relying party to verify the origin of the authenticator with an attestation certificate authority. the information contained in the attestation may thus disclose some information about the user (e.g. which device they are using). ...

PublicKeyCredentialCreationOptions.excludeCredentials - Web APIs
lue an array whose elements are objects with the following properties: type a string describing type of public-key credential to be created. as of this writing (march 2019), only "public-key" may be used. id a buffersource matching an existing public key credential identifier (publickeycredential.rawid). this identifier is generated during the creation of the publickeycredential instance. transports optional an array of strings describing the possible transports between the client and the authenticator. the value of the strings may be: "usb": the authenticator can be contacted via a removable usb link "nfc": the authenticator may be used over nfc (near field communication) "ble": the authenticator may be used over ble (bluetooth low energy) "internal": the authenticator is specif...

PublicKeyCredentialCreationOptions - Web APIs
w public key credentials for an existing user who already have some. publickeycredentialcreationoptions.authenticatorselection optional an object whose properties are criteria used to filter out the potential authenticators for the creation operation. publickeycredentialcreationoptions.attestation optional a string which indicates how the attestation (for the authenticator's origin) should be transported. publickeycredentialcreationoptions.extensions optional an object with several client extensions' inputs. those extensions are used to request additional processing (e.g. dealing with legacy fido apis credentials, prompting a specific text on the authenticator, etc.). methods none. examples // some examples of cose algorithms const cose_alg_ecdsa_w_sha256 = -7; const cose_alg_ecdsa_w_sha...

RTCDataChannel.close() - Web APIs
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. the sequence of events which occurs in response to this method being called: rtcdatachannel.readystate is set to "closing". a background task is established to handle the remainder of the steps below, and close() returns to the caller. the transport layer deals with any buffered messages; the protocol layer decides whether to send them or discard them. the underlying data transport is closed. the rtcdatachannel.readystate property is set to "closed". if the transport was closed with an error, the rtcdatachannel is sent a networkerror event. a close event is sent to the channel. in firefox, the rtcdatachannel interface was implemented ...

RTCDataChannel: close event - Web APIs
before any further data can be transferred using rtcdatachannel, a new data channel instance must be created. bubbles no cancelable no interface event event handler property rtcdatachannel.onclose examples this example sets up a handler for the close event for the rtcdatachannel named dc; its responsibility in this example is to update user interface elements to reflect that there is no longer an ongoing call, and to allow a new call to be started. ...

RTCDataChannel.onclosing - Web APIs
the rtcdatachannel.onclosing property is an eventhandler which specifies a function to be called by the browser when the closing event is received by the rtcdatachannel. this is a simple event which indicates that the data channel is being closed, that is, rtcdatachannel transitions to "closing" state. for example, after rtcdatachannel.close() was called but the underlying data transport might not have been closed yet. syntax rtcdatachannel.onclosing = function; value a function which the browser will call to handle the closing event. the function receives as its sole input parameter the event itself, as an object of type event. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.oncl...

RTCDataChannel.onopen - Web APIs
syntax rtcdatachannel.onopen = function; value a function which the browser will call to handle the open event. the function receives as its only input parameter the event itself, of type event. example this example adds a new data channel to an existing rtcpeerconnection, mypeerconnection. ...

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. it enables the message input box and send button as well as enabling the disconnect button and disabling the connect button. ...

RTCIceCandidate.tcpType - Web APIs
andidate type. the tcptype field's value is set when the rtcicecandidate() constructor is used. you can't directly set its value; instead, its value is automatically extracted from the candidate a-line, if it's formatted properly. syntax var tcptype = rtcicecandidate.tcptype; value a domstring whose value is one of those defined by the rtcicetcpcandidatetype enumerated type. "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. tcptype is null for udp candidates. example in this example, the candidate's protocol and tcptype are used to adj...

RTCIceCandidate.usernameFragment - Web APIs
ignalmsg, arrives from the signaling server that contains an ice candidate to be added to the rtcpeerconnection. to avoid including candidates obsoleted by an ice restart, we can use code like this: const ssnewcandidate = signalmsg => { let candidate = new rtcicecandidate(signalmsg.candidate); let receivers = pc.getreceivers(); receivers.foreach(receiver => { let parameters = receiver.transport.getparameters(); if (parameters.usernamefragment === candidate.usernamefragment) { return; } }); pc.addicecandidate(candidate) .catch(reporterror); } this walks through the list of the rtcrtpreceiver objects being used to receive ice data, and looks to see if the usernamefragment indicated in the candidate matches any of them. if it does, ssnewcandidate() aborts. otherwi...

RTCIceCandidatePairStats.availableIncomingBitrate - Web APIs
ond and is computed over a 1-second interval. the value returned is calculated by adding up the available bit rate for every rtp stream using the connection described by this candidate pair. the returned value doesn't take into account overhead introduced by underlying protocols, including ip, udp, or tcp. note: the returned value is computed using a method similar—but not identical—to the transport independent application specific maximum (tias) described in rfc 3890: 6.2. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcicecandidatepairstats.availableincomingbitrate' in that specification. candidate recommendation initial specification. ...

RTCIceCandidatePairStats.availableOutgoingBitrate - Web APIs
date pair was once in use, but no longer is. the value returned is calculated by adding up the available bit rate for every rtp stream using the connection described by this candidate pair. the returned value doesn't take into account overhead introduced by underlying protocols, including ip, udp, or tcp. note: the returned value is computed using a method similar—but not identical—to the transport independent application specific maximum (tias) described in rfc 3890: 6.2. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcicecandidatepairstats.availableoutgoingbitrate' in that specification. candidate recommendation initial specification. ...

RTCIceCandidateStats.protocol - Web APIs
the rtcicecandidatestats dictionary's protocol property specifies the protocol the specified candidate would use for communication with the remote peer. syntax protocol = rtcicecandidatestats.protocol; value the value is one of the members of the rtciceprotocol enumerated string type: tcp the candidate, if selected, would use tcp as the transport protocol for its data. the tcptype property provides additional information about the kind of tcp candidate represented by the object. udp the candidate will use the udp transport protocol for its data. this is the preferred protocol for media interactions because of its better performance profile. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcicecandidate...

RTCIceCandidateStats.relayProtocol - Web APIs
cifies the protocol being used by a local ice candidate to communicate with the turn server. the ice protocl being used by the candidate otherwise can be obtained from the protocol property. syntax relayprotocol = rtcicecandidatestats.relayprotocol; value a domstring identifying the protocol being used by the endpoint to communicate with the turn server. the possible values are: tcp tcp (transport control protocol) is being used to communicate with the turn server. tls tls (transport layer security) is being used to communicate with the turn server. udp udp (user datagram protocol) is being used to communicate with the turn server. note: this property is only present on rtcicecandidatestats objects that represent local candidates. specifications specification status ...

RTCIceCandidateStats - Web APIs
by the candidate. priority optional the candidate's priority, corresponding to rtcicecandidate.priority. protocol optional a string specifying the protocol (tcp or udp) used to transmit data on the port. relayprotocol optional a string identifying the protocol used by the endpoint for communicating with the turn server; valid values are tcp, udp, and tls. only present for local candidates. transportid optional a string uniquely identifiying the transport object that was inspected in order to obtain the rtctransportstats associated with the candidate correspondin to these statistics. url optional for local candidates, the url property is the url of the ice server from which the candidate was received. this url matches the one included in the rtcpeerconnectioniceevent object representing th...

RTCIceGathererState - Web APIs
the rtcicegathererstate enumerated type provides the string values which can be returned by an rtcicetransport object's gatheringstate. values "new" the rtcicetransport is newly created and has not yet started to gather ice candidates. "gathering" the transport is in the process of gathering candidates. "complete" the transport has finished gathering ice candidates and has sent the end-of-candidates indicator to the remote device. the transport won't gather any further candidates unless an ice restart occurs, at which point the gathering process starts over from scratch. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicegathererstate' in that specification. candidate recommendation init...

RTCIceParameters.password - Web APIs
the rtciceparameters dictionary's password property specifies the ice password that, in tandem with the usernamefragment, uniquely identifies an ice session for its entire duration. syntax password = rtciceparameters.password; value a domstring containing the password that corresponds to the transport's usernamefragment string specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtciceparameters.password' in that specification. candidate recommendation initial specification. ...

RTCIceParameters.usernameFragment - Web APIs
the rtciceparameters dictionary's usernamefragment property specifies the username fragment ("ufrag") that uniquely identifies the corresponding ice session for the duration of the current ice session. syntax ufrag = rtciceparameters.usernamefragment; value a domstring containing the username fragment that, in tandem with the password, uniquely identify the ice session being used by the transport. the string may be up to 256 characters long. see rtcicecandidate.usernamefragment to learn more about username fragments and their role in a connection. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtciceparameters.usernamefragment' in that specification. candidate recommendation initial specifi...

RTCIceParameters - Web APIs
properties usernamefragment a domstring specifying the value of the ice session's username fragment field, ufrag. password a domstring specifying the session's password string. ...

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. you can learn more about ice roles in choosing a candidate pair in webrtc connectivity. values "controlling" the rtcicetransport object is serving as the controlling agent. "controlled" the transport is the controlled agent. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcicerole' in that specification. candidate recommendation initial definition. ...

RTCInboundRtpStreamStats.nackCount - Web APIs
the nackcount property of the rtcinboundrtpstreamstats dictionary is a numeric value indicating the number of times the receiver sent a nack packet to the sender. a nack (negative acknowledgement, also called "generic nack") packet tells the sender that one or more of the rtp packets it sent were lost in transport. syntax var nackcount = rtcinboundrtpstreamstats.nackcount; value an integer value indicating how many times the receiver sent a nack packet to the sender after detecting that one or more packets were lost during transport. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcinboundrtpstreamstats.nackcount' in that specification. candidate recommendation initial definition. ...

RTCOutboundRtpStreamStats.nackCount - Web APIs
the nackcount property of the rtcoutboundrtpstreamstats dictionary is a numeric value indicating the number of times the rtcrtpsender described by this object received a nack packet from the remote receiver. a nack (negative acknowledgement, also called "generic nack") packet is used by the rtcrtpreceiver to inform the sender that one or more rtp packets it sent were lost in transport. syntax var nackcount = rtcoutboundrtpstreamstats.nackcount; value an integer value indicating how many times the sender received a nack packet from the receiver, indicating the loss of one or more packets. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcoutboundrtpstreamstats.nackcount' in that specification. candidate recommen...

RTCPeerConnection() - Web APIs
nging the size of the ice candidate pool may trigger the beginning of ice gathering. iceservers optional an array of rtciceserver objects, each describing one server which may be used by the ice agent; these are typically stun and/or turn servers. if this isn't specified, the connection attempt will be made with no stun or turn server available, which limits the connection to local peers. icetransportpolicy optional the current ice transport policy; this must be one of the values from the rtcicetransportpolicy enum. if this isn't specified, "all" is assumed. peeridentity optional a domstring which specifies the target peer identity for the rtcpeerconnection. if this value is set (it defaults to null), the rtcpeerconnection will not connect to a remote peer unless it can successfully authent...

RTCPeerConnection.createAnswer() - Web APIs
the rtcsessiondescriptioninit dictionary which contains the sdp answer to be delivered to the other peer. example here is a segment of code taken from the code that goes with the article signaling and video calling. this code comes from the handler for the message sent to carry an offer to another peer across the signaling channel. keep in mind that this is part of the signaling process, the transport layer for which is an implementation detail that's entirely up to you. in this case, a websocket connection is used to send a json message with a type field with the value "video-answer" to the other peer, carrying the answer to the device which sent the offer to connect. the contents of the object being passed to the sendtoserver() function, along with everything else in the promise fulfillment ...

RTCPeerConnection.createOffer() - Web APIs
to be secured, that results in an error. operationerror examining the state of the system to determine resource availability in order to generate the offer failed for some reason. example here we see a handler for the negotiationneeded event which creates the offer and sends it to the remote system over a signaling channel. note: keep in mind that this is part of the signaling process, the transport layer for which is an implementation detail that's entirely up to you. in this case, a websocket connection is used to send a json message with a type field with the value "video-offer" to the other peer. the contents of the object being passed to the sendtoserver() function, along with everything else in the promise fulfillment handler, depend entirely on your design. mypeerconnection.create...

RTCPeerConnection.getConfiguration() - Web APIs
ration 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. syntax var configuration = rtcpeerconnection.getconfiguration(); parameters this method takes no input parameters. return value an rtcconfiguration object describing the rtcpeerconnection's current configuration. example this example adds a new certificate to an active connection if it doesn't already have one in use. let configuration = mypeerconnecti...

RTCPeerConnection: icegatheringstatechange event - Web APIs
ection 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. when the value changes to complete, all of the transports that make up the rtcpeerconnection have finished gathering ice candidates. bubbles no cancelable no interface event event handler onicegatheringstatechange note: while you can determine that ice candidate gathering is complete by watching for icegatheringstatechange events and checking for the value of icegatheringstate to become complete,...

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. if sctp hasn't been negotiated, this value is null. 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...

RTCPeerConnection.setConfiguration() - Web APIs
the most common use case for this method (and even then, probably not a very common use case) is to replace the set of ice servers to be used. two potential scenarios in which this might be done: the rtcpeerconnection was instantiated without specifying any ice servers. ...

RTCPeerConnectionIceEvent.candidate - Web APIs
roperty of the rtcpeerconnectioniceevent interface returns the rtcicecandidate associated with the event. syntax var candidate = event.candidate; value an rtcicecandidate object representing the ice candidate that has been received, or null to indicate that there are no further candidates for this negotiation session. 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. candidate recommendation initial specification. ...

RTCPeerConnectionIceEvent - Web APIs
ectioniceevent. it takes two parameters, the first being a domstring representing the type of the event; the second a dictionary containing the rtcicecandidate it refers to. methods a rtcpeerconnectioniceevent being an event, this event also implements these properties. there is no specific rtcdatachannelevent method. examples 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' in that specification. candidate recommendation initial definition. ...

RTCRtpCapabilities - Web APIs
ec (as identified by the mimetype being "video/h264"), there are other values in the capabilities objects indicating how they're different in some way. the mime type strings used by rtp differ from those used elsewhere. see rfc 3555, section 4 for the complete iana registry of these types. there are three special entries that should always be present, representing underlying components of the transport. those components are: red (redundant audio data) the media type of an red entry may vary due to there being several versions of it, but it will end with red, such as video/red or video/fwdred. the base red standard can be found in rfc 2198. there may be multiple entries for red if different forms are supported; each will have a unique media type in that case. fec (forward error correction) ...

RTCRtpEncodingParameters.maxBitrate - Web APIs
track of media it encodes in terms of bits per second. other parameters may further reduce the bandwidth used by the track; for example, maxframerate will, if set low enough, constrain the bandwidth as well. in addition, there's no guarantee that the network interface can support the specified bandwidth, in which case the actual bandwidth will be lower. this value is computed using the standard transport independent application specific maximum (tias) bandwidth as defined by rfc 3890, section 6.2.2; this is the maximum bandwidth needed without considering protocol overheads from ip, tcp or udp, and so forth. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpencodingparameters.maxbitrate' in that speci...

RTCRtpEncodingParameters - Web APIs
s transmission (a feature by which a phone is turned off or the microphone muted automatically in the absence of voice activity). the value is taken from the enumerated string type rtcdtxstatus. maxbitrate an unsigned long integer indicating the maximum number of bits per second to allow for this encoding. other parameters may further constrain the bit rate, such as the value of maxframerate or transport or physical network limitations. maxframerate a double-precision floating-point value specifying the maximum number of frames per second to allow for this encoding. ptime an unsigned long integer value indicating the preferred duration of a media packet in milliseconds. this is typically only relevant for audio encodings. the user agent will try to match this as well as it can, but there is n...

RTCRtpParameters - Web APIs
it is extended separately for senders and receivers in the form of the rtcrtpsendparameters and rtcrtpreceiveparameters dictionaries. to obtain the parameters of a sender or receiver, call its getparameters() method: getparameters() getparameters() properties codecs an array of rtcrtpcodecparameters objects describing the set of codecs from which the sender or receiver will choose. this parameter cannot be changed once initially set. headerextensions an array of zero or more rtp header extensions, each identifying an extension supported by the sender or receiver. ...

RTCRtpReceiver - Web APIs
the rtcrtpreceiver interface of the webrtc api manages the reception and decoding of data for a mediastreamtrack on an rtcpeerconnection. properties rtcrtpreceiver.track read only returns the mediastreamtrack associated with the current rtcrtpreceiver instance. rtcrtpreceiver.transport read only returns the rtcdtlstransport instance over which the media for the receiver's track is received. obsolete properties rtcptransport this property has been removed; the rtp and rtcp transports have been combined into a single transport. use the transport property instead. methods rtcrtpreceiver.getcontributingsources() returns an array of rtcrtpcontributingsource instances for each unique csrc (contributing source) identifier received by the current rtcrtpreceiver in the last ten seco...

RTCRtpSendParameters.encodings - Web APIs
s transmission (a feature by which a phone is turned off or the microphone muted automatically in the absence of voice activity). the value is taken from the enumerated string type rtcdtxstatus. maxbitrate an unsigned long integer indicating the maximum number of bits per second to allow for this encoding. other parameters may further constrain the bit rate, such as the value of maxframerate or transport or physical network limitations. maxframerate a double-precision floating-point value specifying the maximum number of frames per second to allow for this encoding. ptime an unsigned long integer value indicating the preferred duration of a media packet in milliseconds. this is typically only relevant for audio encodings. the user agent will try to match this as well as it can, but there is n...

RTCRtpSender.setStreams() - Web APIs
o 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. for each sender, if the sender's track is non-null and its transport's state is connected, we call setstreams() to add the track to the stream specified. specification...

RTCRtpStreamStats.nackCount - Web APIs
the nackcount property of the rtcrtpstreamstats dictionary is a numeric value indicating the number of times the receiver sent a nack packet to the sender. a nack (negative acknowledgement, also called "generic nack") packet tells the sender that one or more of the rtp packets it sent were lost in transport. syntax var nackcount = rtcrtpstreamstats.nackcount; value an integer value indicating how many times the receiver sent a nack packet to the sender after detecting that one or more packets were lost during transport. note: this value is only available on the receiver. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcinboundrtpstreamstats: nackcount' in that specification. candidate recommendation init...

RTCRtpStreamStats - Web APIs
kind property. previously called mediatype. ssrc the 32-bit integer which identifies the source of the rtp packets this rtcrtpstreamstats object covers. this value is generated per the rfc 3550 specification. trackid a domstring which uniquely identifies the rtcmediastreamtrackstats object representing the associated mediastreamtrack. this is not the same as the value of mediastreamtrack.id. transportid a domstring uniquely identifying the object which was inspected to produce the rtctransportstats object associated with this rtp stream. obsolete fields mediatype renamed to kind in the specification in february 2018. see browser compatibility in rtcrtpstreamstats.kind to determine when browsers made the transition. local-only measurements these properties are computed locally, and a...

Establishing a connection: The WebRTC perfect negotiation pattern - Web APIs
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. after the first deployments of webrtc-capable browsers, it was realized that parts of the negotiation process were more complicated than they needed to be for typical use cases. ...

Using DTMF with WebRTC - Web APIs
echange event with a tone property specifying which tone finished playing, which is an opportunity to update interface elements, for example. when the tone buffer is empty, indicating that all the tones have been sent, a tonechange event with its tone property set to "" (an empty string) is delivered to the connection object. if you'd like to know more about how this works, read rfc 3550: rtp: a transport protocol for real-time applications and rfc 4733: rtp payload for dtmf digits, telephony tones, and telephony signals. the details of how dtmf payloads are handled on rtp are beyond the scope of this article. instead, we'll focus on how to use dtmf within the context of an rtcpeerconnection by studying how an example works. simple example this simple example constructs two rtcpeerconnections, e...

Movement, orientation, and motion: A WebXR example - Web APIs
e. the s key slides the viewer backward move_distance. the a key slides the viewer to the left by move_distance. the d key slides the viewer to the right by move_distance. the up arrow key, ↑, moves the viewer upward by move_distance. the down arrow key, ↓, moves the viewer downward by move_distance. the r key resets the viewer to their starting position and orientation by resetting the transport offsets all to 0. these offsets will be applied by the renderer starting with the next frame drawn. pitching and yawing with the mouse we also have a mousedown event handler which checks to see if the right mouse button is down, and if so, calls the rotateviewby() function, defined next, to calculate and store the new pitch (looking up and down) and yaw (looking left and right) values. funct...

Web Authentication API - Web APIs
50 ]).buffer }, }; // register / create a new credential navigator.credentials.create(createcredentialdefaultargs) .then((cred) => { console.log("new credential", cred); // normally the credential ids available for an account would come from a server // but we can just copy them from above... ...

How to check the security state of an XMLHTTPRequest over SSL - Web APIs
es unused } function dumpsecurityinfo(xhr, error) { let channel = xhr.channel; try { dump("connection status:\n"); if (!error) { dump("\tsucceeded\n"); } else { dump("\tfailed: " + error.name + "\n"); } let secinfo = channel.securityinfo; // print general connection security state dump("security information:\n"); if (secinfo instanceof ci.nsitransportsecurityinfo) { secinfo.queryinterface(ci.nsitransportsecurityinfo); dump("\tsecurity state of connection: "); // check security state flags if ((secinfo.securitystate & ci.nsiwebprogresslistener.state_is_secure) == ci.nsiwebprogresslistener.state_is_secure) { dump("secure connection\n"); } else if ((secinfo.securitystate & ci.nsiwebprogresslistene...

XMLHttpRequest.getAllResponseHeaders() - Web APIs
ated by crlf, or null if no response has been received. if a network error happened, an empty string is returned. 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). these are essentially delimiters separating each of the headers. note: in modern browsers, the header names are returned in all lower case, as per the latest spec. example this example e...

Live streaming web audio and video - Developer guides
al time messaging protocol (rtmp) is a proprietary protocol developed by macromedia (now adobe) and supported by the adobe flash plugin. rtmp comes in various flavours including rtmpe (encrypted), rtmps (secure over ssl/tls) and rtmpt (encapsulated within http requests). rtsp real time streaming protocol (rtsp) controls media sessions between endpoints and is often used together with real-time transport protocol (rtp) and with real-time control protocol (rtcp) for media stream delivery. using rtp with rtcp allows for adaptive streaming. this is not yet supported natively in most browsers, but be aware that firefox os 1.3 supports rtsp. note: some vendors implement propriety transport protocols, such as realnetworks and their real data transport (rdt). rtsp 2.0 rtsp 2.0 is currently in develo...

Setting up adaptive streaming media sources - Developer guides
or media source extensions, see the latest dash.js reference player. hls encoding http live streaming (hls) is an http-based media streaming protocol implemented by apple. it's incorporated into ios and osx platforms and works well on mobile and desktop safari and most android devices with some caveats. media is usually encoded as mpeg-4 (h.264 video and aac audio) and packaged into an mpeg-2 transport stream, which is then broken into segments and saved as one or more .ts media files. apple provides tools to convert media files to the appropriate format. hls encoding tools ther are a number of useful tools available for hls encoding for hls live stream encoding adobe provide a media encoder for mac. the stream segmenter — provided by apple for mac platforms — takes a media stream fro...

Common MIME types - HTTP
ormat (rtf) application/rtf .sh bourne shell script application/x-sh .svg scalable vector graphics (svg) image/svg+xml .swf small web format (swf) or adobe flash document application/x-shockwave-flash .tar tape archive (tar) application/x-tar .tif .tiff tagged image file format (tiff) image/tiff .ts mpeg transport stream video/mp2t .ttf truetype font font/ttf .txt text, (generally ascii or iso 8859-n) text/plain .vsd microsoft visio application/vnd.visio .wav waveform audio format audio/wav .weba webm audio audio/webm .webm webm video video/webm .webp webp image image/webp .woff web open ...

Content Security Policy (CSP) - HTTP
ed to be used; for example (and ideally, from a security standpoint), a server can specify that all content must be loaded using https. a complete data transmission security strategy includes not only enforcing https for data transfer, but also marking all cookies with the secure attribute and providing automatic redirects from http pages to their https counterparts. sites may also use the strict-transport-security http header to ensure that browsers connect to them only over an encrypted channel. using csp configuring content security policy involves adding the content-security-policy http header to a web page and giving it values to control what resources the user agent is allowed to load for that page. for example, a page that uploads and displays images could allow images from anywhere, but r...

Connection management in HTTP/1.x - HTTP
in its infancy, http used a single model to handle such connections. ...

CSP: upgrade-insecure-requests - HTTP
, the latter is effectively a no-op. it is recommended to set either directive, but not both, unless you want to force https on older browsers that do not force it after a redirect to http. the upgrade-insecure-requests directive will not ensure that users visiting your site via links on third-party sites will be upgraded to https for the top-level navigation and thus does not replace the strict-transport-security (hsts) header, which should still be set with an appropriate max-age to ensure that users are not subject to ssl stripping attacks. syntax content-security-policy: upgrade-insecure-requests; examples // header content-security-policy: upgrade-insecure-requests; // meta tag <meta http-equiv="content-security-policy" content="upgrade-insecure-requests"> with the above header set on ...

Index - HTTP
rom the server to the user agent, but has been deprecated by the specification. use set-cookie instead. 102 sourcemap http, http header, reference, response header, header the sourcemap http response header links generated code to a source map, enabling the browser to reconstruct the original source and present the reconstructed original in the debugger. 103 strict-transport-security hsts, http, https, security, header the http strict-transport-security response header (often abbreviated as hsts) lets a web site tell browsers that it should only be accessed using https, instead of using http. 104 te http, reference, header the te request header specifies the transfer encodings the user agent is willing to accept. (you could informally c...

Keep-Alive - HTTP
ers directives parameters a comma-separated list of parameters, each consisting of an identifier and a value separated by the equal sign ('='). the following identifiers are possible: timeout: indicating the minimum amount of time an idle connection has to be kept opened (in seconds). note that timeouts longer than the tcp timeout may be ignored if no keep-alive tcp message is set at the transport level. max: indicating the maximum number of requests that can be sent on this connection before closing it. unless 0, this value is ignored for non-pipelined connections as another request will be sent in the next response. an http pipeline can use it to limit the pipelining. examples a response containing a keep-alive header: http/1.1 200 ok connection: keep-alive content-encoding: gz...

Upgrade - HTTP
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. http/2 explicitly disallows the use of this mechanism/header; it is specific to http/1.1. header type request header, response header forbidden header name yes overview the upgrade header field may be used by clients to invite a server to switch to one (or more) of the listed protocols, in descending preference order. ...

HTTP Messages - HTTP
vides http/1.x messages into frames which are embedded in a stream. data and header frames are separated, this allows header compression. several streams can be combined together, a process called multiplexing, allowing more efficient underlying tcp connections. http frames are now transparent to web developers. this is an additional step in http/2, between http/1.1 messages and the underlying transport protocol. no changes are needed in the apis used by web developers to utilize http frames; when available in both the browser and the server, http/2 is switched on and used. conclusion http messages are the key in using http; their structure is simple, and they are highly extensible. the http/2 framing mechanism adds a new intermediate layer between the http/1.x syntax and the underlying transp...

Proxy servers and tunneling - HTTP
roxy itself (not about the client connecting to it), the via header can be used. via added by proxies, both forward and reverse proxies, and can appear in the request headers and the response headers. http tunneling tunneling transmits private network data and protocol information through public network by encapsulating the data. http tunneling is using a protocol of higher level (http) to transport a lower level protocol (tcp). the http protocol specifies a request method called connect. it starts two-way communications with the requested resource and can be used to open a tunnel. this is how a client behind an http proxy can access websites using ssl (i.e. https, port 443). note, however, that not all proxy servers support the connect method or limit it to port 443 only. see also the htt...

HTTP
etween web browsers and web servers, but it can also be used for other purposes. http follows a classical client-server model, with a client opening a connection to make a request, then waiting until it receives a response. http is a stateless protocol, meaning that the server does not keep any data (state) between two requests. though often based on a tcp/ip layer, it can be used on any reliable transport layer, that is, a protocol that doesn't lose messages silently like udp does. rudp — the reliable update of udp — is a suitable alternative. tutorials learn how to use http with guides and tutorials. overview of http the basic features of the client-server protocol: what it can do and its intended uses. http cache caching is very important for fast web sites. this article describes ...

JSON.stringify() - JavaScript
key and potentially an object as value), you will have to keep track of the iteration count (if it is beyond the first iteration, it is a genuine empty string key). example replacer, as a function function replacer(key, value) { // filtering out properties if (typeof value === 'string') { return undefined; } return value; } var foo = {foundation: 'mozilla', model: 'box', week: 45, transport: 'car', month: 7}; json.stringify(foo, replacer); // '{"week":45,"month":7}' example replacer, as an array if replacer is an array, the array's values indicate the names of the properties in the object that should be included in the resulting json string. json.stringify(foo, ['week', 'month']); // '{"week":45,"month":7}', only keep "week" and "month" properties the space argument the space...

Web video codec guide - Web media technologies
hdr support no variable frame rate (vfr) support no browser compatibility feature chrome edge firefox internet explorer opera safari mpeg-2 support no no no no no yes container support mpeg, mpeg-ts (mpeg transport stream), mp4, quicktime rtp / webrtc compatible no supporting/maintaining organization mpeg / itu specification https://www.itu.int/rec/t-rec-h.262 https://www.iso.org/standard/61152.html licensing proprietary; all patents have expired worldwide with the exception of in malaysia and the philippines as of april 1, 2019, so mpeg-2 can be used free...

Codecs used by WebRTC - Web media technologies
users might prefer. see the article handling media support issues in web content to learn more about how to offer support for your preferred codecs while still being able to fall back on browsers that don't implement that codec. security implications there are interesting potential security issues that come come up while selecting and configuring codecs. webrtc video is protected using datagram transport layer security (dtls), but it is theoretically possible for a motivated party to infer the amount of change that's occurring from frame to frame when using variable bit rate (vbr) codecs, by monitoring the stream's bit rate and how it changes over time. this could potentially allow a bad actor to infer something about the content of the stream, given the ebb and flow of the bit rate. for more ab...

Securing your site - Web security
gest algorithms (such as md5 and sha). this article showcases best practices to use when storing passwords. content security properly configuring server mime types there are several ways incorrect mime types can cause potential security problems with your site. this article explains some of those and shows how to configure your server to serve files with the correct mime types. http strict transport security the strict-transport-security: http header lets a website specify that it may only be accessed using https. http access control the cross-origin resource sharing standard provides a way to specify what content may be loaded from other domains. you can use this to prevent your site from being used improperly; in addition, you can use it to establish resources that other sites are expre...

Types of attacks - Web security
tion is available on a sub-domain, this mechanism can be abused in a session fixation attack. when the user visits a page on the parent domain (or another subdomain), the application may trust the existing value sent in the user's cookie. this could allow an attacker to bypass csrf protection or hijack a session after the user logs in. alternatively, if the parent domain does not use http strict-transport-security with includesubdomains set, a user subject to an active mitm (perhaps connected to an open wifi network) could be served a response with a set-cookie header from a non-existent sub-domain. the end result would be much the same, with the browser storing the illegitimate cookie and sending it to all other pages under example.com. session fixation should primarily be mitigated by regenerat...