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RTCRtpSender - Web APIs
the rtcrtpsender interface provides the ability to control and obtain details about how a particular mediastreamtrack is encoded and sent to a remote peer.
...you can also obtain access to an rtcdtmfsender which can be used to send dtmf codes (to simulate the user pressing buttons on a telephone's dial pad) to the remote peer.
... properties rtcrtpsender.dtmf read only an rtcdtmfsender which can be used to send dtmf tones using telephone-event payloads on the rtp session represented by the rtcrtpsender object.
...And 10 more matches
RTCRtpSender.setParameters() - Web APIs
the setparameters() method of the rtcrtpsender interface applies changes the configuration of sender's track, which is the mediastreamtrack for which the rtcrtpsender is responsible.
... syntax var promise = rtcrtpsender.setparameters(parameters) parameters parameters an object conforming with the rtcrtpsendparameters dictionary, specifying options for the rtcrtpsender; these include potential codecs that could be use for encoding the sender's track.
... return value a promise that resolves when the rtcrtpsender.track property is updated with the given parameters.
...And 5 more matches
RTCPeerConnection.getSenders() - Web APIs
the rtcpeerconnection method getsenders() returns an array of rtcrtpsender objects, each of which represents the rtp sender responsible for transmitting one track's data.
... a sender object provides methods and properties for examining and controlling the encoding and transmission of the track's data.
... syntax var senders = rtcpeerconnection.getsenders(); return value an array of rtcrtpsender objects, one for each track on the connection.
...And 4 more matches
RTCRtpSender.getCapabilities() static function - Web APIs
the static function rtcrtpsender.getcapabilities() returns an rtcrtpcapabilities object describing the codecs and capabilities supported by the rtcrtpsender.
... syntax let rtpcapabilities = rtcrtpsender.getcapabilities(kind); parameters kind a domstring indicating the type of media for which you wish to get the sender's capability to receive.
... description as a static function, this is always called using the form: capabilities = rtcrtpsender.getcapabilities("audio"); the returned set of capabilities is the most optimistic possible list.
...And 4 more matches
RTCRtpSender.setStreams() - Web APIs
the rtcrtpsender method setstreams() associates the sender's track with the specified mediastream or array of mediastream objects.
... syntax rtcrtpsender.setstreams(mediastream); rtcrtpsender.setstreams([mediastream...]); parameters mediastream or [mediastream...] optional an mediastream object—or an array of multiple mediastream objects—identifying the streams to which the rtcrtpsender's track belongs.
... exceptions invalidstateerror the sender's connection is closed.
...And 4 more matches
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.
...And 4 more matches
RTCDTMFSender - Web APIs
the rtcdtmfsender interface provides a mechanism for transmitting dtmf codes on a webrtc rtcpeerconnection.
... you gain access to the connection's rtcdtmfsender through the rtcrtpsender.dtmf property on the audio track you wish to send dtmf with.
...fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">eventtarget</text></a><polyline points="111,25 121,20 121,30 111,25" stroke="#d4dde4" fill="none"/><line x1="121" y1="25" x2="151" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/rtcdtmfsender" target="_top"><rect x="151" y="1" width="130" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="216" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">rtcdtmfsender</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} properties rtcdtmfsender.tone...
...And 3 more matches
RTCRtpSender.replaceTrack() - Web APIs
the rtcrtpsender method replacetrack() replaces the track currently being used as the sender's source with a new mediastreamtrack.
... syntax trackreplacedpromise = sender.replacetrack(newtrack); parameters newtrack optional a mediastreamtrack specifying the track with which to replace the rtcrtpsender's current source track.
... if newtrack was omitted or was null, replacetrack() simply stops the sender.
...And 3 more matches
RTCRtpSender.getParameters() - Web APIs
the getparameters() method of the rtcrtpsender interface returns an rtcrtpsendparameters object describing the current configuration for the encoding and transmission of media on the sender's track.
... syntax var rtpsendparameters = rtpsender.getparameters() parameters none.
... return value an rtcrtpsendparameters object indicating the current configuration of the sender.
...And 2 more matches
RTCRtpSender.getStats() - Web APIs
the rtcrtpsender method getstats() asynchronously requests an rtcstatsreport object which provides statistics about outgoing traffic on the rtcpeerconnection which owns the sender, returning a promise which is fulfilled when the results are available.
... syntax var promise = rtcrtpsender.getstats(); return value a javascript promise which is fulfilled once the statistics are available.
... the returned rtcstatsreport accumulates the statistics for all of the streams being sent using the rtcrtpsender, as well as the statistics for any dependencies those streams have.
...And 2 more matches
nsISyncMessageSender
idl file: mozilla-central/source/dom/base/nsimessagemanager.idl inherits from: nsimessagelistenermanager message "senders" have a single "other side" to which messages are sent.
...see message manager interfaces for more details on the distinction between message senders and message broadcasters.
... [optional] in jsval objects, [optional] in nsiprincipal principal); jsval sendrpcmessage([optional] in astring messagename, [optional] in jsval obj, [optional] in jsval objects, [optional] in nsiprincipal principal); sendsyncmessage() like sendasyncmessage(), except blocks the sender until all listeners of the message have been invoked.
... throws ns_error_not_initialized if the sender is not initialized.
RTCDTMFSender.insertDTMF() - Web APIs
the insertdtmf() method on the rtcdtmfsender interface starts sending dtmf tones to the remote peer over the rtcpeerconnection.
... sending of the tones is performed asynchronously, with tonechange events sent to the rtcdtmfsender every time a tone starts or ends.
... syntax rtcdtmfsender.insertdtmf(tones[, duration[, intertonegap]]); parameters tones a domstring containing the dtmf codes to be transmitted to the recipient.
... example tbd specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdtmfsender.insertdtmf()' in that specification.
RTCRtpSender.dtmf - Web APIs
WebAPIRTCRtpSenderdtmf
the read-only dtmf property on the rtcrtpsender interface returns a rtcdtmfsender object which can be used to send dtmf tones over the rtcpeerconnection .
... see using dtmf for details on how to make use of the returned rtcdtmfsender object.
... syntax var dtmfsender = rtcrtpsender.dtmf; value an rtcdtmfsender which can be used to send dtmf over the rtp session, or null if the track being carried by the rtp session or the rtcpeerconnection as a whole doesn't support dtmf.
... only audio tracks can support dtmf, and typically only one audio track per rtcpeerconnection will have an associated rtcdtmfsender example tbd specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpsender.dtmf' in that specification.
nsIMessageSender
idl file: mozilla-central/source/dom/base/nsimessagemanager.idl inherits from: nsimessagelistenermanager message senders enable clients to send asynchronous messages to a single "other side".
...see message manager interfaces for more details on the distinction between message senders and message broadcasters.
... throws ns_error_not_initialized if the sender is not initialized.
RTCDTMFSender.toneBuffer - Web APIs
the rtcdtmfsender interface's tonebuffer property returns a string containing a list of the dtmf tones currently queued for sending to the remote peer over the rtcpeerconnection.
... syntax var tonebuffer = rtcdtmfsender.tonebuffer; value a domstring listing the tones to be played.
... example tbd specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdtmfsender.tonebuffer' in that specification.
RTCRtpSender.track - Web APIs
the track read-only property of the rtcrtpsender interface returns the mediastreamtrack which is being handled by the rtcrtpsender.
... syntax var mediastreamtrack = rtcrtpsender.track value a mediastreamtrack object representing the media associated with the rtcrtpsender.
... if no track is associated with the sender, this value is null, in which case the sender transmits nothing.
RTCRtpTransceiver.sender - Web APIs
the read-only sender property of webrtc's rtcrtptransceiver interface indicates the rtcrtpsender responsible for encoding and sending outgoing media data for the transceiver's stream.
... syntax var rtpsender = rtcrtptransceiver.sender; value an rtcrtpsender object used to encode and send media whose media id matches the current value of mid.
... specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtptransceiver.sender' in that specification.
RTCDTMFSender.ontonechange - Web APIs
the ontonechange property of the rtcdtmfsender interface is used to set the event handler for the tonechange event, which is sent to the rtcdtmfsender each time a tone begins or ends.
... syntax rtcdtmfsender.ontonechange = tonechangehandlerfunction; value a function which is called when a tonechange event is sent to the rtcdtmfsender, indicating that a dtmf tone has either started playing, or if all tones have finished playing.
RTCDTMFSender: tonechange event - Web APIs
the tonechange event is sent to an rtcdtmfsender by the webrtc api to indicate when dtmf tones previously queued for sending (by calling rtcdtmfsender.insertdtmf()) begin and end.
... this can be done using addeventlistener(): dtmfsender.addeventlistener("tonechange", ev => { let tone = ev.tone; if (tone === "") { tone = "&lt;none&gt;"; } document.getelementbyid("playingtone").innertext = tone; }, false); you can also just set the ontonechange event handler property directly: dtmfsender.ontonechange = function( ev ) { let tone = ev.tone; if (tone === "") { tone = "&lt;none&gt;" } document.getelementby...
Index - Web APIs
WebAPIIndex
3220 rtcdtmfsender audio, dtmf, interface, media, pstn, rtcdtmfsender, reference, telephony, touch-tone, webrtc, webrtc api listen to these events using addeventlistener() or by assigning an event listener to the oneventname property of this interface.
... 3221 rtcdtmfsender.insertdtmf() api, dtmf, media, method, rtcdtmfsender, reference, telephony, touch-tone, webrtc, webrtc api, insertdtmf the insertdtmf() method on the rtcdtmfsender interface starts sending tones to the remote peer over the rtcpeerconnection.
... sending of the tones is performed asynchronously, with tonechange events sent to the rtcdtmfsender every time a tone starts or ends.
...And 49 more matches
Using DTMF with WebRTC - Web APIs
when you wish to transmit dtmf signals, you first need to decide which track to send them on, since dtmf is sent as a series of out-of-band payloads on the rtcrtpsender responsible for transmitting that track's data to the other peer.
... once the track is selected, you can obtain from its rtcrtpsender the rtcdtmfsender object you'll use for sending dtmf.
... from there, you can simply call rtcdtmfsender.insertdtmf() to enqueue dtmf signals to be sent on the track to the other peer.
...And 14 more matches
RTCPeerConnection.addTrack() - Web APIs
syntax rtpsender = rtcpeerconnection.addtrack(track, stream...); parameters track a mediastreamtrack object representing the media track to add to the peer connection.
... return value the rtcrtpsender object which will be used to transmit the media data.
... note: every rtcrtpsender is paired with an rtcrtpreceiver to make up an rtcrtptransceiver.
...And 10 more matches
Codecs used by WebRTC - Web media technologies
the sender is not required to support this mechanism, however, so you have to be prepared to receive media at a different resolution than you requested.
... an audio sender is never required to use discontinuous transmission or comfort noise.
...the most efficient way is to use the static method rtcrtpsender.getcapabilities() (or the equivalent rtcrtpreceiver.getcapabilities() for a receiver), specifying the type of media as the input parameter.
...And 9 more matches
RTCOutboundRtpStreamStats - Web APIs
the rtcoutboundrtpstreamstats dictionary is the rtcstats-based object which provides metrics and statistics related to an outbound rtp stream being sent by an rtcrtpsender.
... fircount an integer value which indicates the total number of full intra request (fir) packets which this rtcrtpsender has sent to the remote rtcrtpreceiver.
... nackcount an integer value indicating the total number of negative acknolwedgement (nack) packets this rtcrtpsender has received from the remote rtcrtpreceiver.
...And 8 more matches
Introduction to the Real-time Transport Protocol (RTP) - Web APIs
these correspond to the following three types of transport supported by rtcpeerconnection: rtcrtpsender rtcrtpsenders handle the encoding and transmission of mediastreamtrack data to a remote peer.
... the senders for a given connection can be obtained by calling rtcpeerconnection.getsenders().
... rtcrtptransceiver an rtcrtptransceiver is a pair of one rtp sender and one rtp receiver which share an sdp mid attribute, which means they share the same sdp media m-line (representing a bidirectional srtp stream).
...And 6 more matches
Introduction to Public-Key Cryptography - Archive of obsolete content
the sender encrypts, or scrambles, information before sending it.
... authentication allows the recipient of information to determine its origin-that is, to confirm the sender's identity.
... nonrepudiation prevents the sender of information from claiming at a later date that the information was never sent.
...And 4 more matches
WebRTC Statistics API - Web APIs
the webrtc api has a vast array of statistics available, covering the entire breadth of the webrtc connectivity system, from sender to receiver and peer to peer.
... rtcicecandidatestats rtcstats media-source statistics about the media produced by the mediastreamtrack attached to an rtp sender.
... rtcsctptransportstats rtcstats sender statistics related to a specific rtcrtpsender and the media associated with that sender.
...And 4 more matches
nsIMsgHeaderParser
astring makefulladdress( in astring aname, in astring aaddress ); parameters aname the name of the sender.
... aaddress the address of the sender.
... string makefulladdressstring( in string aname, in string aaddress ); parameters aname the name of the sender.
...And 3 more matches
RTCDtlsTransport - Web APIs
the rtcdtlstransport interface provides access to information about the datagram transport layer security (dtls) transport over which a rtcpeerconnection's rtp and rtcp packets are sent and received by its rtcrtpsender and rtcrtpreceiver objects.
... when not using bundle when the connection is created without using bundle, each rtp or rtcp component of each rtcrtptransceiver has its own rtcdtlstransport; that is, every rtcrtpsender and rtcrtpreceiver, has its own transport, and all rtcdatachannel objects share a transport dedicated to sctp.
...if your code accesses rtcrtpsenders and/or rtcrtpreceivers directly, you may encounter situations where they're initially separate, then half or more of them get closed and the senders and receivers updated to refer to the appropriate remaining rtcdtlstransport objects.
...And 3 more matches
RTCRtpEncodingParameters - Web APIs
an instance of the webrtc api's rtcrtpencodingparameters dictionary describes a single configuration of a codec for an rtcrtpsender.
... it's used in the rtcrtpsendparameters describing the configuration of an rtp sender's encodings; rtcrtpdecodingparameters is used to describe the configuration of an rtp receiver's encodings.
...that is, for rtp senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data.
...And 3 more matches
RTCStatsReport - Web APIs
the rtcstatsreport interface provides a statistics report obtained by calling one of the rtcpeerconnection.getstats(), rtcrtpreceiver.getstats(), and rtcrtpsender.getstats() methods.
...the rtcrtpreceiver and rtcrtpsender versions of getstats() specifically only return statistics available to the incoming or outgoing stream on which you call them.
... sender an object containing statistics about the rtcrtpsender for a stream on the rtcpeerconnection.
...And 3 more matches
IPDL Tutorial
their values are serialized by the sender and deserialized by the receiver.
... in ipdl, there are three different semantics: asynchronous semantics; the sender is not blocked.
...we call this synchronous semantics, as the sender blocks until the receiver receives the message and sends back a reply.
...And 2 more matches
RTCInboundRtpStreamStats.pliCount - Web APIs
the plicount property of the rtcinboundrtpstreamstats dictionary states the number of times the rtcrtpreceiver described by these statistics sent a picture loss indication (pli) packet to the sender.
... syntax var plicount = rtcinboundrtpstreamstats.plicount; value an integer value indicating the number of times a pli packet was sent by the rtcrtpreceiver to the sender.
... these are sent by the receiver's decoder to notify the encoder (the sender) that an undefined amount of coded video data, which may span frame boundaries, has been lost.
...And 2 more matches
RTCOutboundRtpStreamStats.firCount - Web APIs
the fircount property of the rtcoutboundrtpstreamstats dictionary indicates the number of full intra request (fir) that the remote rtcrtpreceiver has sent to this rtcrtpsender.
... a fir packet is sent when the receiver finds that it has fallen behind and needs to skip frames in order to catch up; the sender should respond by sending a full frame instead of a delta frame.
... syntax var fircount = rtcoutboundrtpstreamstats.fircount; value an integer value indicating how many fir packets have been received by the sender during the current connection.
...And 2 more matches
RTCOutboundRtpStreamStats.pliCount - Web APIs
the plicount property of the rtcoutboundrtpstreamstats dictionary states the number of times the remote peer's rtcrtpreceiver sent a picture loss indication (pli) packet to the rtcrtpsender for which this object provides statistics.
... syntax var plicount = rtcoutboundrtpstreamstats.plicount; value an integer value indicating the number of times a pli packet was sent to this sender by the remote peer's rtcrtpreceiver.
... these are sent by the receiver's decoder to notify the sender's encoder that an undefined amount of coded video data, which may span frame boundaries, has been lost.
...And 2 more matches
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.
... headerextensions an array of zero or more rtp header extensions, each identifying an extension supported by the sender or receiver.
...And 2 more matches
RTCRtpSendParameters.encodings - Web APIs
the rtcrtpsendparameters dictionary's encodings property is an rtcrtpencodingparameters object providing configuration settings for the encoder being used for the rtcrtpsender's track.
...that is, for rtp senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data.
... codecpayloadtype when describing a codec for an rtcrtpsender, codecpayloadtype is a single 8-bit byte (or octet) specifying the codec to use for sending the stream; the value matches one from the owning rtcrtpparameters object's codecs parameter.
...And 2 more matches
RTCRtpStreamStats.pliCount - Web APIs
the plicount property of the rtcrtpstreamstats dictionary states the number of times the stream's receiving end sent a picture loss indication (pli) packet to the sender.
... syntax var plicount = rtcrtpstreamstats.plicount; value an integer value indicating the number of times a pli packet was sent by the stream's receiver to the sender.
... a pli message is used by video decoders (running on the receiving end of the stream) to notify the encoder (the sender) that an undefined amount of coded video data, which may span frame boundaries, has been lost.
...And 2 more matches
RTCRtpStreamStats - Web APIs
the rtcrtpstreamstats dictionary is returned by the rtcpeerconnection.getstats(), rtcrtpsender.getstats(), and rtcrtpreceiver.getstats() methods to provide detailed statistics about webrtc connectivity.
... fircount a count of the total number of full intra request (fir) packets received by the sender.
... nackcount the number of times the receiver notified the sender that one or more rtp packets has been lost by sending a negative acknowledgement (nack, also called "generic nack") packet to the sender.
...And 2 more matches
window.postMessage() - Web APIs
if you do expect to receive messages from other sites, always verify the sender's identity using the origin and possibly source properties.
... any window (including, for example, http://evil.example.com) can send a message to any other window, and you have no guarantees that an unknown sender will not send malicious messages.
...popup.postmessage("hello there!", "http://example.com"); function receivemessage(event) { // do we trust the sender of this message?
...And 2 more matches
Message manager overview
ame script into this browser frame (tab) sendasyncmessage() : send a message to all frame scripts in this browser frame addmessagelistener() : start listening to a specific message from frame scripts removemessagelistener() : stop listening to a specific message interfaces nsiprocesschecker nsiframescriptloader nsimessagelistenermanager nsimessagesender how to access the browser message manager can be accessed as a property of the xul <browser> element: // chrome script let browsermm = gbrowser.selectedbrowser.messagemanager; content process content frame message manager description there's a content frame message manager for every open tab.
...ts : access privileged objects and apis addeventlistener() : listen to dom events addmessagelistener() : receive messages from the chrome process sendasyncmessage() : send asynchronous messages to the chrome process sendsyncmessage() : send synchronous messages to the chrome process interfaces nsidomeventtarget nsimessagelistenermanager nsimessagesender nsisyncmessagesender nsicontentframemessagemanager how to access the content frame message manager is the global object in frame scripts.
...ript() : load a process script into the content process broadcastasyncmessage() : send a message to process scripts addmessagelistener() : start listening to a specific message from process scripts removemessagelistener() : stop listening to a specific message interfaces nsiprocesschecker nsiprocessscriptloader nsimessagelistenermanager nsimessagesender how to access you can access a ppmm using the getchildat() function in the gppmm: // parent process let ppmm = services.ppmm.getchildat(1); child process child process message manager description there's one child process message manager (cpmm) in each child process.
... its most important attributes and functions are: components : access privileged objects and apis addmessagelistener() : receive messages from the parent process sendasyncmessage() : send asynchronous messages to the parent process sendsyncmessage() : send synchronous messages to the parent process interfaces nsimessagelistenermanager nsimessagesender nsisyncmessagesender nsicontentprocessmessagemanager how to access code running in a child process can access the cpmm with code like this: // child process script let cpmm = cc["@mozilla.org/childprocessmessagemanager;1"] .getservice(ci.nsisyncmessagesender); you can also access it as the cpmm property of services.jsm, if you are in the child ...
WebChannel.jsm
parameters callback callback function containing function(id, message, sendercontext) parameters.
... id - webchannel id of the incoming messages message - incoming message object sendercontext - incoming message context - this should be treated as an opaque object and passed to the .send() method stoplistening() resets the callback for messages on this channel.
... parameters message the message object that will be sent sendercontext the sendercontext parameter passed to the .listen method.
... examples setting up a webchannel between chrome code and a webpage chrome code let channel = new webchannel(webchannelid, services.io.newuri("https://mozilla.org", null, null)); // receive messages channel.listen(function (webchannelid, message, sendercontext) { // send messages channel.send({ data: { greeting: true } }, sendercontext); }); webpage code receive messages from an existing webchannel in content code window.addeventlistener("webchannelmessagetocontent", function(e) { // receive messages console.log(e.detail); }, true); send messages to an existing webchannel in chrome code window.dispatchevent(new window.customevent("webchannelmessagetochrome", { detail: { id: webchannelid, message: { something: true } } })); ...
Activity Manager examples
as a result, all junk processing activities (assuming that we process accounts in parallel) dealing with messages coming from the same sender will be grouped together.
....constructor("@mozilla.org/activity-process;1", "nsiactivityprocess", "init"); const nsactevent = components.constructor("@mozilla.org/activity-event;1", "nsiactivityevent", "init"); // step 1: implement nsiactivitycontextdisplayhelper to show a // customized display text for our context type function sendercontextdisplayhelper() { // user stuff here..
... } sendercontextdisplayhelper.prototype = { getcontextdisplaytext: function(contexttype, contextobj) { // in this particular example we know that contexttype is "sender" // since we also pass the contexttype along with the contextobject // in some cases, one helper can be registered for a group of context types // we know that the context object is the author of the message // localization is omitted return "messages coming from " + contextobj.surname + ", " + contextobj.firstname; } } // step 2: register the helper for this context type gactivitymanager.registercontextdisplayhelper("sender", new sendercontextdisplayhelper()); // step 3: create the process ...
... let process = new nsactprocess("processing folder: " + folder.prettiestname, gjunkprocessor); process.cancelhandler = new canceljunkprocess(); // folder is being filtered/processed process.addsubject(folder); process.contexttype = "sender"; // assuming msg is an instance of nsimsghdr process.contextobj = msg.author; ...
RTCInboundRtpStreamStats.firCount - Web APIs
the fircount property of the rtcinboundrtpstreamstats dictionary indicates the number of full intra request (fir) packets have been sent by the receiver to the sender.
... syntax var fircount = rtcinboundrtpstreamstats.fircount; value an integer value indicating how many fir packets have been received by the sender during the current connection.
... the receiver sends a fir packet to the sender any time it falls bahind or loses packets and cannot decode the incoming stream any longer because of the lost data.
... this tells the sender to send a full frame instead of a delta frame, so that the receiver can catch up.
RTCPeerConnection.removeTrack() - Web APIs
the rtcpeerconnection.removetrack() method tells the local end of the connection to stop sending media from the specified track, without actually removing the corresponding rtcrtpsender from the list of senders as reported by rtcpeerconnection.getsenders().
... if the track is already stopped, or is not in the connection's senders list, this method has no effect.
... syntax pc.removetrack(sender); parameters mediatrack a rtcrtpsender specifying the sender to remove from the connection.
... var pc, sender; navigator.getusermedia({video: true}, function(stream) { pc = new rtcpeerconnection(); var track = stream.getvideotracks()[0]; sender = pc.addtrack(track, stream); }); document.getelementbyid("closebutton").addeventlistener("click", function(event) { pc.removetrack(sender); pc.close(); }, false); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.removetrack()' in that specification.
RTCRtpStreamStats.firCount - Web APIs
the fircount property of the rtcrtpstreamstats dictionary indicates the number of full intra request (fir) packets have been sent by the receiver to the sender.
... syntax var fircount = rtcrtpstreamstats.fircount; value an integer value indicating how many fir packets have been received by the sender during the current connection.
... the receiver sends a fir packet to the sender any time it falls bahind or loses packets and cannot decode the incoming stream any longer because of the lost data.
... this tells the sender to send a full frame instead of a delta frame, so that the receiver can catch up.
RTCRtpTransceiver.currentDirection - Web APIs
each describes how the transceiver's associated rtcrtpsender and rtcrtpreceiver behave as shown in the table below.
... value rtcrtpsender behavior rtcrtpreceiver behavior "sendrecv" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... "sendonly" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... [1] to determine if a sender has at least one active encoding, the user agent gets its parameters using rtcrtpsender.getparameters(), then looks at the parameters' encodings property; if any of the listed encodings has its active property set to true, the sender has an active encoding.
RTCRtpTransceiver.direction - Web APIs
each describes how the transceiver's associated rtcrtpsender and rtcrtpreceiver behave as shown in the table below.
... value rtcrtpsender behavior rtcrtpreceiver behavior "sendrecv" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... "sendonly" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... [1] to determine if a sender has at least one active encoding, the user agent gets its parameters using rtcrtpsender.getparameters(), then looks at the parameters' encodings property; if any of the listed encodings has its active property set to true, the sender has an active encoding.
RTCRtpTransceiver.setCodecPreferences() - Web APIs
that means that either the associated rtcrtpsender or the rtcrtpreceiver needs to support every codec in the list.
... to determine which codecs are supported by the transceiver, call the sender's getcapabilities() and the receiver's getcapabilities() methods and get the codecs list from the results of each.
... the following code snippet demonstrates how to get both the list of codecs supported by the transceiver's rtcrtpsender and rtcrtpreceiver.
... var availsendcodecs = transceiver.sender.getcapabilities("video").codecs; var availreceivecodecs = transceiver.receiver.getcapabilities("video").codecs; specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtptransceiver.setcodecpreferences()' in that specification.
RTCRtpTransceiver - Web APIs
the webrtc interface rtcrtptransceiver describes a permanent pairing of an rtcrtpsender and an rtcrtpreceiver, along with some shared state.
... sender read only the rtcrtpsender object responsible for encoding and sending data to the remote peer.
... stopped indicates whether or not sending and receiving using the paired rtcrtpsender and rtcrtpreceiver has been permanently disabled, either due to sdp offer/answer, or due to a call to stop().
...the associated sender stops sending data, and the associated receiver likewise stops receiving and decoding incoming data.
RTCRtpTransceiverDirection - Web APIs
each describes how the transceiver's associated rtcrtpsender and rtcrtpreceiver behave as shown in the table below.
... value rtcrtpsender behavior rtcrtpreceiver behavior "sendrecv" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... "sendonly" offers to send rtp data, and will do so if the other peer accepts the connection and at least one of the sender's encodings is active1.
... [1] to determine if a sender has at least one active encoding, the user agent gets its parameters using rtcrtpsender.getparameters(), then looks at the parameters' encodings property; if any of the listed encodings has its active property set to true, the sender has an active encoding.
RTCStatsType - Web APIs
sender an object containing statistics about the rtcrtpsender for a stream on the rtcpeerconnection.
... if kind is "audio", this object is of type rtcaudiosenderstats; if kind is "video", this is an rtcvideosenderstats object.
... 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.
WebRTC API - Web APIs
rtcrtpsender manages the encoding and transmission of data for a mediastreamtrack on an rtcpeerconnection.
... interfaces rtcdtmfsender manages the encoding and transmission of dual-tone multi-frequency (dtmf) signaling for an rtcpeerconnection.
... events tonechange either a new dtmf tone has begun to play over the connection, or the last tone in the rtcdtmfsender's tonebuffer has been sent and the buffer is now empty.
... using dtmf with webrtc webrtc's support for interacting with gateways that link to old-school telephone systems includes support for sending dtmf tones using the rtcdtmfsender interface.
Index
MozillaTechXPCOMIndex
741 nsimessagelistener this interface is used to receive messages sent using the nsimessagebroadcaster, nsimessagesender, or nsisyncmessagesender interfaces.
...to access this service, use: 743 nsimessagesender message senders enable clients to send asynchronous messages to a single "other side".
... 993 nsisyncmessagesender message "senders" have a single "other side" to which messages are sent.
nsIProcessScriptLoader
if this function is called on a chromemessagesender: it will load the process script only into this chromemessagesender's child process aallowdelayedload should always be true.
... it's possible that the child process for this chromemessagesender is not yet available at the time of the loadprocessscript() call, and if you pass false in this situation, your process script won't be loaded.
... if you are calling loadprocessscript() on a chromemessagesender, you should always pass true here.
nsISmsService
to create an instance, use: var smsservice = components.classes["@mozilla.org/sms/smsservice;1"] .createinstance(components.interfaces.nsismsservice); method overview [implicit_jscontext] nsidommozsmsmessage createsmsmessage(in long id, in domstring delivery, in domstring sender, in domstring receiver, in domstring body, in jsval timestamp, in bool read ); unsigned short getnumberofmessagesfortext(in domstring text); boolean hassupport(); void send(in domstring number, in domstring message, in long requestid, [optional] in unsigned long long processid); methods createsmsmessage() [implicit_jscontext] nsidommozsmsmessage createsmsmessag...
...e( in long id, in domstring delivery, in domstring sender, in domstring receiver, in domstring body, in jsval timestamp, in bool read ); parameters id a number representing the id of the message.
... sender a domstring with the name of the sender.
RTCDtlsTransport.state - Web APIs
examples this example presents a function, tallysenders(), which iterates over an rtcpeerconnection's rtcrtpsenders, tallying up how many of them are in various states.
... the function returns an object containing properties whose values indicate how many of the senders are in each state.
...*/ function tallysenders(pc) { let results = { transportmissing: 0, connectionpending: 0, connected: 0, closed: 0, failed: 0, unknown: 0 }; let senderlist = pc.getsenders(); senderlist.foreach(sender => { let transport = sender.transport; if (!transport) { results.transportmissing++; } else { switch(transport.state) { case "new": case "connecting": results.connectionpending++; break; case "connected": results.connected++; break; case "closed": results.closed++; break; case "failed": results.failed++; break; default: results.unknown++...
RTCIceTransport: gatheringstatechange event - Web APIs
bubbles no cancelable no interface event event handler property ongatheringstatechange the key difference between gatheringstatechange and icegatheringstatechange is that the latter represents the overall state of the connection including every rtcicetransport used by every rtcrtpsender and every rtcrtpreceiver on the entire connection.
... examples this example creates a handler for gatheringstatechange events on each rtcrtpsender associated with a given rtcpeerconnection.
... here, the addeventlistener() method is called to add a listener for gatheringstatechange events: pc.getsenders().foreach(sender => { sender.transport.icetransport.addeventlistener("gatheringstatechange", ev => { let transport = ev.target; if (transport.gatheringstate === "complete") { /* this transport has finished gathering candidates, but others may still be working on it */ } }, false); likewise, you can use the ongatheringstatechange event handler property: pc.getsenders().foreach(sender => { sender.transport.icetransport.ongatheringstatechange = ev => { let transport = ev.target; if (transport.gatheringstate === "complete") { /* this transport has finished gathering candidates, but others may still be working on it */ } }; }); specifi...
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.
RTCInboundRtpStreamStats.sliCount - Web APIs
the slicount property of the rtcinboundrtpstreamstats dictionary indicates how many slice loss indication (sli) packets the rtcrtpreceiver for which this object provdes statistics sent to the remote rtcrtpsender.
... in general, what's usually of interest is that the higher this number is, the more the stream data is becoming corrupted between the sender and the receiver, requiring resends or dropping frames.
... syntax var slicount = rtcinboundrtpstreamstats.slicount; value an unsigned integer indicating the number of sli packets this receiver sent to the remote sender due to lost runs of macroblocks.
RTCInboundRtpStreamStats - Web APIs
fircount an integer value which indicates the total number of full intra request (fir) packets which this receiver has sent to the sender.
... plicount an integer specifying the number of times the receiver has notified the sender that some amount of encoded video data for one or more frames has been lost, using picture loss indication (pli) packets.
... slicount an integer indicating the number of times the receiver sent a slice loss indication (sli) frame to the sender to tell it that one or more consecutive (in terms of scan order) video macroblocks have been lost or corrupted.
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.
RTCOutboundRtpStreamStats.sliCount - Web APIs
the slicount property of the rtcoutboundrtpstreamstats dictionary indicates how many slice loss indication (sli) packets the rtcrtpsender received from the remote rtcrtpreceiver for the rtp stream described by this object.
... an sli packet is used by a decoder to let the encoder (the sender) know that it's detected corruption of one or more consecutive macroblocks, in scan order, in the received media.in general, what's usually of interest is that the higher this number is, the more the stream data is becoming corrupted between the sender and the receiver, causing the receiver to request retransmits or to drop frames entirely.
... syntax var slicount = rtcoutboundrtpstreamstats.slicount; value an unsigned integer indicating the number of sli packets the sender received from the receiver due to lost runs of macroblocks.
RTCPeerConnection - Web APIs
each rtp receiver manages the reception and decoding of data for a mediastreamtrack on an rtcpeerconnectiongetsenders()the rtcpeerconnection method getsenders() returns an array of rtcrtpsender objects, each of which represents the rtp sender responsible for transmitting one track's data.getstats()the rtcpeerconnection method getstats() returns a promise which resolves with data providing statistics about either the overall connection or about the specified mediastreamtrack.getstreambyid() the rtcpeerconnectio...
...because this method has been deprecated, you should instead use removetrack() if your target browser versions have implemented it.removetrack()the rtcpeerconnection.removetrack() method tells the local end of the connection to stop sending media from the specified track, without actually removing the corresponding rtcrtpsender from the list of senders as reported by rtcpeerconnection.getsenders().restartice()the webrtc api's rtcpeerconnection interface offers the restartice() method to allow a web application to easily request that ice candidate gathering be redone on both ends of the connection.setconfiguration() the rtcpeerconnection.setconfiguration() method sets the current configuration of the rtcpeerconnection b...
... rtcpeerconnection.createdtmfsender() creates a new rtcdtmfsender, associated to a specific mediastreamtrack, that will be able to send dtmf phone signaling over the connection.
RTCRemoteOutboundRtpStreamStats.reportsSent - Web APIs
the rtcremoteoutboundrtpstreamstats dictionary's reportssent property provides the number of sender reports (srs) the remote peer has transmitted to the local peer.
... syntax let reportcount = rtcremoteoutboundrtpstreamstats.reportssent; value an integer value which indicates the total number of rtcp sender reports so far sent by the remote peer to the local peer.
... usage notes sender reports, described in rfc 3550, section 6.4.1 with an overview in rfc 3550, section 6.4, are used by rtp to share data transmission quality feedback between the two peers.
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.
RTCRtpStreamStats.sliCount - Web APIs
the slicount property of the rtcrtpstreamstats dictionary indicates how many slice loss indication (sli) packets were received by the sender.
... syntax var slicount = rtcrtpstreamstats.slicount; value an unsigned long integer indicating the number of sli packets the sender received from the receiver due to lost runs of macroblocks.
... note: this value is sent by the sender to the receiver and is only present for video media.
Index - HTTP
WebHTTPHeadersIndex
82 index http, http header, index found 115 pages: 83 keep-alive general header, http, http header, reference the keep-alive general header allows the sender to hint about how the connection may be used to set a timeout and a maximum amount of requests.
... 107 trailer http, reference, header the trailer response header allows the sender to include additional fields at the end of chunked messages in order to supply metadata that might be dynamically generated while the message body is sent, such as a message integrity check, digital signature, or post-processing status.
...it is used for tracking message forwards, avoiding request loops, and identifying the protocol capabilities of senders along the request/response chain.
Want-Digest - HTTP
the header contains identifiers for one or more digest algorithms that the sender wishes the responder to use to create the digest.
... the sender may use quality values to indicate its preference ordering among the choices it offers.
... examples want-digest: sha-256 want-digest: sha-512;q=0.3, sha-256;q=1, md5;q=0 basic operation the sender provides a list of digests which it is prepared to accept, and the server uses one of them: request: get /item want-digest: sha-256;q=0.3, sha;q=1 response: http/1.1 200 ok digest: sha-256=x48e9qookqqrvdts8nojrjn3owduoywxbf7kbu9dbpe= unsupported digests the server does not support any of the requested digest algorithms, so uses a different algorithm: request: get /item want-di...
HTTP Index - HTTP
WebHTTPIndex
161 index http, http header, index found 122 pages: 162 keep-alive general header, http, http header, reference the keep-alive general header allows the sender to hint about how the connection may be used to set a timeout and a maximum amount of requests.
... 188 trailer http, reference, header the trailer response header allows the sender to include additional fields at the end of chunked messages in order to supply metadata that might be dynamically generated while the message body is sent, such as a message integrity check, digital signature, or post-processing status.
...it is used for tracking message forwards, avoiding request loops, and identifying the protocol capabilities of senders along the request/response chain.
ui/frame - Archive of obsolete content
in this case you should pass the frame's url property as the targetorigin: frame.postmessage("ping", frame.url); on(event, listener) assign a listener to a frame event: var { frame } = require("sdk/ui/frame"); var frame = new frame({ url: "./frame.html" }); frame.on("message", pong) function pong(e) { if (e.data == "ping") { // message only the sender, and not any frames attached to other browser windows e.source.postmessage("pong", "*"); } } parameters event : string the name of the event to listen to.
... var { frame } = require("sdk/ui/frame"); var frame = new frame({ url: "./frame.html" }); frame.once("message", pong) function pong(e) { if (e.data == "ping") { // message only the sender, and not any frames attached to other browser windows e.source.postmessage("pong", "*"); } } parameters event : string the name of the event to listen to.
popChallengeResponse - Archive of obsolete content
[note that a different -- integer must be used for each challenge.] sender generalname, -- the name of the sender.
... -- request is being made) of rand, where rand is specified as -- rand ::= sequence { -- int integer, -- - the randomly-generated integer a (above) -- senderhash octet string -- - the result of applying the one-way function (owf) to -- - the sender's general name -- } -- the size of "int" must be small enough such that "rand" can be -- contained within a single pkcs #1 encryption block.
Trees - Archive of obsolete content
ArchiveMozillaXULTutorialTrees
there might be multiple columns, such as the sender and the subject.
... example tree with treechildren example 2 : source view <tree flex="1"> <treecols> <treecol id="sender" label="sender" flex="1"/> <treecol id="subject" label="subject" flex="2"/> </treecols> <treechildren> <treeitem> <treerow> <treecell label="joe@somewhere.com"/> <treecell label="top secret plans"/> </treerow> </treeitem> <treeitem> <treerow> <treecell label="mel@whereever.com"/> <treecell label="let's do lunch"/> </treerow> </treeitem> </treechildren> </tree> as can be seen in the image, the tree has been created with two rows of data.
RDF in Fifty Words or Less - Archive of obsolete content
formatting a graph into xml: <rdf:rdf xmlns:rdf="http://www.w3.org/tr/wd-rdf-syntax#" xmlns:sm="http://www.mozilla.org/smart-mail/schema#"> <rdf:description about="http://www.mozilla.org/smart-mail/get-mail.cgi?user=waterson&folder=inbox"> <sm:message id="4025293"> <sm:recipient> chris waterson "waterson@netscape.com" </sm:recipient> <sm:sender> aunt helga "helga@netcenter.net" </sm:sender> <sm:received-by>x-wing.mcom.com</sm:received-by> <sm:subject>great recipe for yam soup!</sm:subject> <sm:body> http://www.mozilla.org/smart-mail/get-body.cgi?id=4025293 </sm:body> </sm:message> <sm:message id="4025294"> <sm:recipient> chris waterson "waterson@nets...
...cape.com" </sm:recipient> <sm:sender> sarah waterson "waterson.2@postbox.acs.ohio-state.edu" </sm:sender> <sm:received-by>x-wing.mcom.com</sm:received-by> <sm:subject>we won our ultimate game</sm:subject> <sm:body> http://www.mozilla.org/smart-mail/get-body.cgi?id=4025294 </sm:body> </sm:message> </rdf:description> </rdf:rdf> upon receipt of the above monstrosity, the rdf engine folds the rdf into the graph at the appropriate place, and the tree control that actually implements the ui to the bookmarks is notified that it should begin drawing some icons for the latest message about yam soup from aunt helga.
Marking up a letter - Learn web development
inline semantics the names of the sender and receiver (and tel and email) should be marked up with strong importance.
... the first address and first date in the letter should have a class attribute value of sender-column.
Message manager overview
it's a chromemessagesender object, which implements the following interfaces: nsiprocesschecker nsiframescriptloader nsimessagelistenermanager nsimessagesender the browser message manager can be accessed as a property of the xul <browser> element: // chrome script let browsermm = gbrowser.selectedbrowser.messagemanager; process message managers process message managers correspond to process boundaries, and enabl...
... // child process script let childprocessmessagemanager = cc["@mozilla.org/childprocessmessagemanager;1"] .getservice(ci.nsisyncmessagesender); parent process message manager the parent process message manager lives in the implements nsimessagelistenermanager nsimessagebroadcaster child process message manager implements nsimessagelistenermanager nsimessagesender loading scripts per child process since the process message managers do not support script loading the only way to implement per-child process code is to us...
IME handling guide
then, mozilla::imecontentobserver::imenotificationsender tries to send the pending notifications when it might become safe to do that.
... notify_ime_of_text_change, notify_ime_of_selection_change, notify_ime_of_position_change and notify_ime_of_composition_event_handled are always sent by following order: notify_ime_of_text_change notify_ime_of_selection_change notify_ime_of_position_change notify_ime_of_composition_event_handled if sending one of above notifications causes higher priority notification, the sender should abort to send remaning notifications and restart from highet priority notification again.
nsIContentFrameMessageManager
idl file: mozilla-central/source/dom/base/nsimessagemanager.idl inherits from: nsisyncmessagesender this interface provides the environment for scripts that are loaded into content frames using the nsiframescriptloader interface.
... frame scripts can send either synchronous or asynchronous messages to the chrome process: for details on these messaging apis see the documentation for the nsicontentframemessagemanager's parent classes nsisyncmessagesender and nsimessagesender.
nsIFrameMessageManager
you should use nsimessagelistenermanager or nsimessagesender instead.
... see also content process event handling nsiframemessagelistener nsisyncmessagesender ...
nsIFrameScriptLoader
if this function is called on a chromemessagesender: it will load the frame script only into this chromemessagesender's frame aallowdelayedload should always be true.
... it's possible that the frame for this chromemessagesender is not yet available at the time of the loadframescript() call, and if you pass false in this situation, your process script won't be loaded.
nsIMessageBroadcaster
see message manager interfaces for more details on the distinction between message senders and message broadcasters.
... throws ns_error_not_initialized if the sender is not initialized.
nsIMessageListener
idl file: mozilla-central/source/dom/base/nsimessagemanager.idl inherits from: nsisupports this interface is used to receive messages sent using the nsimessagebroadcaster, nsimessagesender, or nsisyncmessagesender interfaces.
... objects an object whose properties are any cross process object wrappers exposed by the sender as the third argument to the message-sending function.
nsIMsgFolder
displayrecipients boolean readonly: should we be displaying recipients instead of the sender.
... for example, in the sent folder, recipients are more relevant than the sender.
nsISmsDatabaseService
to create an instance, use: var smsservice = components.classes["@mozilla.org/sms/smsdatabaseservice;1"] .createinstance(components.interfaces.nsismsdatabaseservice); method overview long savereceivedmessage(in domstring asender, in domstring abody, in unsigned long long adate); long savesentmessage(in domstring areceiver, in domstring abody, in unsigned long long adate); void getmessage(in long messageid, in long requestid, [optional] in unsigned long long processid); void deletemessage(in long messageid, in long requestid, [optional] in unsigned long long processid); void createmessagelist(in nsidommozsmsfilter...
...estid, [optional] in unsigned long long processid); void getnextmessageinlist(in long listid, in long requestid, [optional] in unsigned long long processid); void clearmessagelist(in long listid); void markmessageread(in long messageid, in boolean value, in long requestid, [optional] in unsigned long long processid) methods savereceivedmessage() void savereceivedmessage( in domstring asender, in domstring abody, in unsigned long long adate ); parameters asender a domstring with the sender of the text message.
XPCOM Interface Reference
leservicensilogininfonsiloginmanagernsiloginmanagercryptonsiloginmanageriemigrationhelpernsiloginmanagerprompternsiloginmanagerstoragensiloginmetainfonsimimeinputstreamnsimacdocksupportnsimarkupdocumentviewernsimemorynsimemorymultireporternsimemorymultireportercallbacknsimemoryreporternsimemoryreportermanagernsimenuboxobjectnsimessagebroadcasternsimessagelistenernsimessagelistenermanagernsimessagesendernsimessagewakeupservicensimessengernsimicrosummarynsimicrosummarygeneratornsimicrosummaryobservernsimicrosummaryservicensimicrosummarysetnsimimeconverternsimimeheadersnsimodulensimsgaccountnsimsgaccountmanagerextensionnsimsgcompfieldsnsimsgcustomcolumnhandlernsimsgdbhdrnsimsgdbviewnsimsgdbviewcommandupdaternsimsgdatabasensimsgfilternsimsgfiltercustomactionnsimsgfilterlistnsimsgfoldernsimsgheaderpa...
...portsarraynsisupportscstringnsisupportscharnsisupportsdoublensisupportsfloatnsisupportsidnsisupportsinterfacepointernsisupportsprboolnsisupportsprint16nsisupportsprint32nsisupportsprint64nsisupportsprtimensisupportspruint8nsisupportspruint16nsisupportspruint32nsisupportspruint64nsisupportsprimitivensisupportsprioritynsisupportsstringnsisupportsvoidnsisupportsweakreferencensisyncjpakensisyncmessagesendernsitxttohtmlconvnsitaggingservicensitaskbarpreviewnsitaskbarpreviewbuttonnsitaskbarpreviewcontrollernsitaskbarprogressnsitaskbartabpreviewnsitaskbarwindowpreviewnsitelemetrynsitextinputprocessornsitextinputprocessorcallbacknsitextinputprocessornotificationnsithreadnsithreadeventfilternsithreadinternalnsithreadmanagernsithreadobservernsithreadpoolnsithreadpoollistenernsitimernsitimercallbacknsitool...
BroadcastChannel: message event - Web APIs
bubbles no cancelable no interface messageevent event handler property onmessage examples live example in this example there's a "sender" <iframe> that broadcasts the contents of a <textarea> when the user clicks a button.
... sender <h1>sender</h1> <label for="message">type a message to broadcast:</label><br/> <textarea id="message" name="message" rows="1" cols="40">hello</textarea> <button id="broadcast-message" type="button">broadcast message</button> body { border: 1px solid black; padding: .5rem; height: 150px; font-family: "fira sans", sans-serif; } h1 { font: 1.6em "fira sans", sans-serif; margin-bottom: 1rem; } textarea { padding: .2rem; } label, br { margin: .5rem 0; } button { vertical-align: top; height: 1.5rem; } const channel = new broadcastchannel('example-channel'); const messagecontrol = document.queryselector('#message'); const broadcas...
RTCDTMFToneChangeEvent.tone - Web APIs
if all queued tones have finished playing (that is, rtcdtmfsender.tonebuffer is empty).
... dtmfsender.ontonechange = function( ev ) { let tone = ev.tone; if (tone === "") { tone = "&lt;none&gt;" } document.getelementbyid("playingtone").innertext = tone; }; specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdtmftonechangeevent.tone' in that specification.
RTCIceCandidatePair.local - Web APIs
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.
RTCIceCandidatePair.remote - Web APIs
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.
RTCIceTransport.getLocalCandidates() - Web APIs
example this simple example gets the local candidate list from the rtcicetransport for the first rtcrtpsender on the rtcpeerconnection, then outputs to the console all of the candidates in the list.
... 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.
RTCIceTransport.getRemoteCandidates() - Web APIs
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.
RTCIceTransport: selectedcandidatepairchange event - Web APIs
let icetransport = pc.getsenders[0].transport.icetransport; let localprotoelem = document.getelementbyid("local-protocol"); let remoteprotoelem = document.getelementbyid("remote-protocol"); icetransport.addeventlistener("selectedcandidatepairchange", ev => { let pair = icetransport.getselectedcandidatepair(); localprotoelem.innertext = pair.local.protocol.touppercase(); remoteprotoelem.innertext = pair.remote.protocol.to...
... let icetransport = pc.getsenders[0].transport.icetransport; let localprotoelem = document.getelementbyid("local-protocol"); let remoteprotoelem = document.getelementbyid("remote-protocol"); icetransport.onselectedcandidatepairchange = ev => { let pair = icetransport.getselectedcandidatepair(); localprotoelem.innertext = pair.local.protocol.touppercase(); remoteprotoelem.innertext = pair.remote.protocol.touppercase(); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'selectedcandidatepairchange' in that specification.
RTCInboundRtpStreamStats.remoteId - Web APIs
the remoteid property of the rtcinboundrtpstreamstats dictionary specifies the id of the rtcremoteoutboundrtpstreamstats object representing the remote peer's rtcrtpsender which is sending the media to the local peer.
... syntax var remotestatsid = rtcinboundrtpstreamstats.remoteid; value a domstring containing the id of the rtcremoteoutboundrtpstreamstats object that represents the remote peer's rtcrtpsender for the synchronization source represented by this stats object.
RTCOutboundRtpStreamStats.framesEncoded - Web APIs
the framesencoded property of the rtcoutboundrtpstreamstats dictionary indicates the total number of frames that have been encoded by this rtcrtpsender for this media source.
... syntax var framesencoded = rtcoutboundrtpstreamstats.framesencoded; value an integer value indicating the total number of video frames that this sender has encoded so far for this stream.
RTCOutboundRtpStreamStats.trackId - Web APIs
the trackid property of the rtcoutboundrtpstreamstats dictionary indicates the id of the rtcsenderaudiotrackattachmentstats or rtcsendervideotrackattachmentstats object representing the mediastreamtrack which is being sent on this stream.
... syntax var trackstatsid = rtcoutboundrtpstreamstats.trackid; value a domstring containing the id of the rtcsenderaudiotrackattachmentstats or rtcsendervideotrackattachmentstats object representing the track which is the source of the media being sent on this stream.
RTCPeerConnection.addTransceiver() - Web APIs
each transceiver represents a bidirectional stream, with both an rtcrtpsender and an rtcrtpreceiver associated with it.
... sendencodings optional a list of encodings to allow when sending rtp media from the rtcrtpsender.
RTCRemoteOutboundRtpStreamStats - Web APIs
reportssent an integer value indicating the total number of rtcp sender report (sr) blocks that this ssrc has sent.
... usage notes the rtcremoteoutboundrtpstreamstats object's remotetimestamp proeprty provides statistics based on the received data's ntp timestamp taken from an rtcp sender report (sr) block.
RTCRtcpParameters - Web APIs
it's used as the value of the rtcp property of the parameters of an rtcrtpsender or rtcrtpreceiver.
... examples this example obtains the canonical name (cname) being used for rtcp on an rtcrtpsender or rtcrtpreceiver.
RTCRtpCapabilities - Web APIs
the rtcrtpcapabilities dictionary is a data type used to describe the capabilities of an rtcrtpsender or rtcrtpreceiver in response to a call to the rtcrtpsender.getcapabilities() or rtcrtpreceiver.getcapabilities() static functions, both of which return an array of rtcrtpcapabilities objects.
... properties codecs an array of rtcrtpcodeccapability objects, each describing one of the codecs supported by the rtcrtpsender or rtcrtpreceiver.
RTCRtpTransceiver.stop() - Web APIs
the stop() method in the rtcrtptransceiver interface permanently stops the transceiver by stopping both the associated rtcrtpsender and rtcrtpreceiver.
... usage notes when you call stop() on a transceiver, the sender immediately stops sending media and each of its rtp streams are closed using the rtcp "bye" message.
RTCRtpTransceiver.stopped - Web APIs
the read-only stopped property on the rtcrtptransceiver interface indicates whether or not the transceiver's associated sender and receiver have both been stopped.
... syntax var isstopped = rtcrtptransceiver.stopped; value a boolean value which is true if the transceiver's sender will no longer send data, and its receiver will no longer receive data.
RTCTrackEvent.transceiver - Web APIs
the transceiver pairs the track's receiver with an rtcrtpsender.
... syntax var rtptransceiver = trackevent.transceiver; value the rtcrtptransceiver which pairs the receiver with a sender and other properties which establish a single bidirectional srtp stream for use by the track associated with the rtctrackevent.
RTCTrackEventInit.transceiver - Web APIs
the transceiver pairs the track's receiver with an rtcrtpsender to allow bidirectional communication.
... syntax var trackeventinit = { receiver: rtpreceiver, track: mediastreamtrack, streams: [videostream], transceiver: rtptransceiver }; var rtptransceiver = trackeventinit.transceiver; value the rtcrtptransceiver which pairs the receiver with a sender and other properties which establish a single bidirectional srtp stream for use by the track associated with the rtctrackevent.
Signaling and video calling - Web APIs
name the sender's username.
... sdp the sdp (session description protocol) string describing the local end of the connection from the perspective of the sender (or the remote end of the connection from the receiver's point of view).
Web APIs
WebAPI
ermissions photocapabilities plugin pluginarray point pointerevent popstateevent 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 rtcicetransportstat...
...cnetworktype rtcofferansweroptions rtcofferoptions rtcoutboundrtpstreamstats rtcpeerconnection rtcpeerconnectioniceerrorevent rtcpeerconnectioniceevent 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 readablestreambyobr...
HTTP headers - HTTP
WebHTTPHeaders
trailer allows the sender to include additional fields at the end of chunked message.
...if the upgrade header field is specified, then the sender must also send the connection header field with the upgrade option specified.
Communicating using "port" - Archive of obsolete content
to send messages from one side to the other, use port.emit() to receive messages sent from the other side, use port.on() messages are asynchronous: that is, the sender does not wait for a reply from the recipient but just emits the message and continues processing.
Content Scripts - Archive of obsolete content
to send messages from one side to the other, use port.emit() to receive messages sent from the other side, use port.on() messages are asynchronous: that is, the sender does not wait for a reply from the recipient but just emits the message and continues processing.
Modifying Web Pages Based on URL - Archive of obsolete content
to send a message from one side to the other, sender calls port.emit() and receiver listens using port.on().
Modifying the Page Hosted by a Tab - Archive of obsolete content
to send a message from one side to the other, sender calls port.emit() and receiver listens using port.on().
Interaction between privileged and non-privileged pages - Archive of obsolete content
node) return; var doc = node.ownerdocument; callback(json.parse(node.nodevalue), doc, function(response) { node.nodevalue = json.stringify(response); var event = doc.createevent("htmlevents"); event.initevent("something-response", true, false); return node.dispatchevent(event); }); }, false, true); }, callback: function(request, sender, callback) { if (request.foo) { return settimeout(function() { callback({bar: 2}); }, 1000); } if (request.baz) { return settimeout(function() { callback({quux: 4}); }, 3000); } if (request.mozilla) { return alert("alert in chrome"); } return callback(null); } } something.listen_request(something.callback); message pass...
Message Summary Database - Archive of obsolete content
this includes a set of per-message flags, the more commonly used headers (e.g., subject, sender, from, to, cc, date, etc), and a few other attributes, e.g., keywords.
RDF Datasource How-To - Archive of obsolete content
by combining statements from the imap datasource and address book datasource, above, we'd be able to identify the sender of "message 126" as an "important friend".
Building Trees - Archive of obsolete content
here is an example of a tree using an sqlite datasource: <tree datasources="profile:messages.sqlite" ref="*" querytype="storage" flags="dont-build-content"> <treecols> <treecol id="subject" label="subject" flex="3"/> <treecol id="sender" label="sender" flex="2"/> <treecol id="date" label="date" flex="1"/> </treecols> <template> <query> select subject, sender, date from messages </query> <action> <treechildren> <treeitem uri="?"> <treerow> <treecell label="?subject"/> <treecell label="?sender"/> <treecell label="?date"/> </treerow> ...
tree - Archive of obsolete content
ArchiveMozillaXULtree
, onselect, rows, seltype, statedatasource, tabindex, treelines properties accessibletype, builderview, columns, contentview, currentindex, disablekeynavigation, disabled, editingcolumn, editingrow, enablecolumndrag, firstordinalcolumn, inputfield, seltype, selstyle, tabindex, treeboxobject, view examples a tree with several columns <tree flex="1" rows="2"> <treecols> <treecol id="sender" label="sender" flex="1"/> <treecol id="subject" label="subject" flex="2"/> </treecols> <treechildren> <treeitem> <treerow> <treecell label="joe@somewhere.com"/> <treecell label="top secret plans"/> </treerow> </treeitem> <treeitem> <treerow> <treecell label="mel@whereever.com"/> <treecell label="let's do lunch"/> </tree...
treechildren - Archive of obsolete content
espace(), isequalnode, issamenode, issupported(), lookupnamespaceuri, lookupprefix, normalize(), queryselector(), queryselectorall(), removeattribute(), removeattributenode(), removeattributens(), removechild(), removeeventlistener(), replacechild(), setattribute(), setattributenode(), setattributenodens(), setattributens(), setuserdata example <tree flex="1"> <treecols> <treecol id="sender" label="sender" flex="1"/> <treecol id="subject" label="subject" flex="2"/> </treecols> <treechildren> <treeitem> <treerow> <treecell label="joe@somewhere.com"/> <treecell label="top secret plans"/> </treerow> </treeitem> <treeitem> <treerow> <treecell label="mel@whereever.com"/> <treecell label="let's do lunch"/> </tree...
Mozilla release FAQ - Archive of obsolete content
if you think you're getting it from mozilla because the x-mailer field mentions mozilla, then you're misinterpreting the meaning of the x-mailer field -- it indicates that the sender is *using* netscape communicator to deliver the mail.
Extentsions FAQ - Archive of obsolete content
friday, september 29 - october 6, 2006 (↑ top) how can we programmatically extract the subject, sender, date and the body of an e- mail when you are reading one (in thunderbird presumably)?
Asynchronous - MDN Web Docs Glossary: Definitions of Web-related terms
for humans, e-mail is an asynchronous communication method; the sender sends an email and the recipient will read and reply to the message when it's convenient to do so, rather than doing so at once.
Packet - MDN Web Docs Glossary: Definitions of Web-related terms
backward error correction is when the receiver requests the sender to retransmit the entire data unit.
Signature (security) - MDN Web Docs Glossary: Definitions of Web-related terms
on receiving the message, the verification process authenticates the sender - uses the sender's public key to decrypt the signature and recover the hash, which can only be created with the sender's private key, and checks message integrity - compares the hash with a newly calculated one from the received document (the two hashes will differ if the document has been tampered with) the system fails if the private key is compromised or the recipient is deceitfully given the wrong public key.
Communicating with frame scripts
objects an object whose properties are any cpows exposed by the sender as the third argument to sendasyncmessage() in the example below the listener just logs all the messages details: // chrome script messagemanager.addmessagelistener("my-addon@me.org:my-e10s-extension-message", listener); function listener(message) { console.log(message.name); console.log(message.sync); console.log(message.data); console.log(message.target); console.log(message.
Message manager
guides message manager overview frame script loading and lifetime communicating with frame scripts performance best practices frame script environment limitations of frame scripts process scripts api reference nsiframescriptloader nsimessagelistenermanager nsimessagebroadcaster nsimessagesender nsisyncmessagesender nsicontentframemessagemanager nsiprocessscriptloader ...
Services.jsm
cache nsicacheservice cache service cache2 nsicachestorageservice cache storage service clipboard nsiclipboard clipboard console nsiconsoleservice error console service contentprefs nsicontentprefservice content preferences service cookies nsicookiemanager2 cookie manager 2 service cpmm nsimessagesender child process message manager4 crashmanager crashmanager.jsm dirsvc nsidirectoryservice nsiproperties directory service domstoragemanager nsidomstoragemanager dom storage manager domrequest nsidomrequestservice domrequest service downloads nsidownloadmanager download manager droppedlinkhandler nsidropp...
AsyncTestUtils extended framework
for example, {count: 1, subject: "my suitcase"} would result in a single message with the subject "my suitcase" with a random sender and random recipient.
NSS_3.11.10_release_notes.html
bug 398680: assertion botch in ssl3_registerserverhelloextensionsender doing second handshake with ssl_forcehandshake bug 403240: threads hanging in nss_initlock bug 403888: memory leak in trustdomain.c bug 416067: certutil -l -h token doesn't report token authentication failure bug 417637: tstclnt crashes if -p option is not specified bug 421634: don't send an sni client hello extension bearing an ipv6 address bug 422918: add verisign class 3 public pri...
nsIMessageListenerManager
dom/base/nsimessagemanager.idlscriptable this interface enables clients to listen for messages sent using the nsimessagebroadcaster, nsimessagesender, or nsisyncmessagesender interfaces.
nsIMessenger
if they want a list suitable for display, they should pass in an ahistory pointer, and they'll get returned an array with strings containing something like subject and sender of the message - other possible info is the folder containing the message, and the preview text, if available.
nsMsgSearchAttrib
*/ [scriptable, uuid(a83ca7e8-4591-4111-8fb8-fd76ac73c866)] interface nsmsgsearchattrib { const nsmsgsearchattribvalue custom = -2; /* a custom term, see nsimsgsearchcustomterm */ const nsmsgsearchattribvalue default = -1; const nsmsgsearchattribvalue subject = 0; /* mail and news */ const nsmsgsearchattribvalue sender = 1; const nsmsgsearchattribvalue body = 2; const nsmsgsearchattribvalue date = 3; const nsmsgsearchattribvalue priority = 4; /* mail only */ const nsmsgsearchattribvalue msgstatus = 5; const nsmsgsearchattribvalue to = 6; const nsmsgsearchattribvalue cc = 7; const nsmsgsearchattribvalue toorcc = 8; const nsmsgsearchattribvalue alladdresses = 9; const ns...
XPCOM Interface Reference by grouping
editabletext nsiaccessibleevent nsiaccessiblehyperlink nsiaccessiblehypertext nsiaccessibleimage nsiaccessibleprovider nsiaccessibleretrieval nsiaccessiblerole nsiaccessiblescrolltype nsiaccessibleselectable nsiaccessiblestates nsiaccessibletable nsiaccessibletext nsiaccessibletreecache nsiaccessiblevalue nsiaccessnode nsisyncmessagesender script nsiscriptableunescapehtml nsiscriptableunicodeconverter nsiscripterror nsiscripterror2 stylesheet nsistylesheetservice url nsiuri nsiurl util nsidomserializer nsidomxpathevaluator nsidomxpathexception nsidomxpathexpression nsidomxpathresult xslt nsixsltexception nsixsltprocessor download ...
nsMsgMessageFlags
senderauthed 0x00000200 partial 0x00000400 indicates whether or not the body we have is a complete message.
Buddy icons in mail
for the message pane, the icon we will show is on disk at: <profile home>/nim/<value of pref aim.session.screenname>/picture/<screenname for sender email address>.gif when trying to determine the screenname for the sender, we search the addressbook that we are using for collection.
Index
for the message pane, the icon we will show is on disk at: <profile home>/nim/<value of pref aim.session.screenname>/picture/<screenname for sender email address>.gif when trying to determine the screenname for the sender, we search the addressbook that we are using for collection.
RTCDTMFToneChangeEvent - Web APIs
dtmfsender.addeventlistener("change", function(event) { if (event.tone !== "") { dialstringbox.innertext += event.tone; } else { senddtmfbutton.disabled = false; } }, false); specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdtmftonechangeevent' in that specification.
RTCDataChannel: error event - Web APIs
examples // strings for each of the sctp cause codes found in rfc // 4960, section 3.3.10: // https://tools.ietf.org/html/rfc4960#section-3.3.10 const sctpcausecodes = [ "no sctp error", "invalid stream identifier", "missing mandatory parameter", "stale cookie error", "sender is out of resource (i.e., memory)", "unable to resolve address", "unrecognized sctp chunk type received", "invalid mandatory parameter", "unrecognized parameters", "no user data (sctp data chunk has no data)", "cookie received while shutting down", "restart of an association with new addresses", "user-initiated abort", "protocol violation" ]; dc.addeventlistener("error", ev => ...
RTCIceCandidatePair - Web APIs
var icetransport = pc.getsenders()[0].transport.icetransport; var localproto = document.getelementbyid("local-protocol"); var remoteproto = document.getelementbyid("remote-protocol"); icetransport.onselectedcandidatepairchange = function(event) { var pair = icetransport.getselectedcandidatepair(); localprotocol.innertext = pair.local.protocol.touppercase(); remoteprotocol.innertext = pair.remote.protocol.touppercase(); }...
RTCIceCandidatePairStats.availableOutgoingBitrate - Web APIs
the returned value is undefined in each of the following situations: the underlying implementation doesn't support computing a sender-side estimate of the outgoing bit rate.
RTCIceTransport.getSelectedCandidatePair() - Web APIs
var icetransport = pc.getsenders()[0].transport.icetransport; var localproto = document.getelementbyid("local-protocol"); var remoteproto = document.getelementbyid("remote-protocol"); icetransport.onselectedcandidatepairchange = function(event) { var pair = icetransport.getselectedcandidatepair(); localprotocol.innertext = pair.local.protocol.touppercase(); remoteprotocol.innertext = pair.remote.protocol.touppercase(); }...
RTCIceTransport.ongatheringstatechange - Web APIs
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.
RTCIceTransport.onselectedcandidatepairchange - Web APIs
var icetransport = pc.getsenders()[0].transport.icetransport; var localproto = document.getelementbyid("local-protocol"); var remoteproto = document.getelementbyid("remote-protocol"); icetransport.onselectedcandidatepairchange = function(event) { var pair = icetransport.getselectedcandidatepair(); localprotocol.innertext = pair.local.protocol.touppercase(); remoteprotocol.innertext = pair.remote.protocol.touppercase(); }...
RTCIceTransport.onstatechange - Web APIs
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.
RTCIceTransport: statechange event - Web APIs
let icetransport = pc.getsenders()[0].transport.icetransport; icetransport.addeventlistener("statechange", ev => { if (icetransport.state === "failed") { handlefailure(pc); } }, false); the same code, using the onstatechange event handler property, looks like this: let icetransport = pc.getsenders()[0].transport.icetransport; icetransport.onstatechange = ev => { if (icetransport.state === "failed") { handlefa...
RTCInboundRtpStreamStats.averageRtcpInterval - Web APIs
because the interval's value is determined in part by the number of active senders, it will be different for each user of a service.
RTCInboundRtpStreamStats.bytesReceived - Web APIs
this value can be used to calculate an approximation of the average media data rate: avgdatarate = rtcinboundrtpstreamstats.bytesreceived / elapsedtime; this value gets reset to zero if the sender's ssrc identifier changes for any reason.
RTCOutboundRtpStreamStats.averageRtcpInterval - Web APIs
because the interval's value is determined in part by the number of active senders, it will be different for each user of a service.
RTCOutboundRtpStreamStats.lastPacketSentTimestamp - Web APIs
the lastpacketsenttimestamp property of the rtcoutboundrtpstreamstats dictionary indicates the time at which the rtcrtpsender described by this rtcoutboundrtpstreamstats object last transmitted a packet to the remote receiver.
RTCOutboundRtpStreamStats.qpSum - Web APIs
the qpsum property of the rtcoutboundrtpstreamstats dictionary is a value generated by adding the quantization parameter (qp) values for every frame this sender has produced to date on the video track corresponding to this rtcoutboundrtpstreamstats object.
RTCPeerConnection.addStream() - Web APIs
said stream was added to: stream.addtrack(track); if (pc.addtrack) { pc.addtrack(track, stream); } else { // if you have code listening for negotiationneeded events: settimeout(() => pc.dispatchevent(new event('negotiationneeded'))); } // remove a track from a stream and the peer connection said stream was added to: stream.removetrack(track); if (pc.removetrack) { pc.removetrack(pc.getsenders().find(sender => sender.track == track)); } else { // if you have code listening for negotiationneeded events: settimeout(() => pc.dispatchevent(new event('negotiationneeded'))); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcpeerconnection.addstream()' in that specification.
RTCPeerConnection.close() - Web APIs
all rtcrtpsender objects are considered to be stopped once this returns (they may still be in the process of stopping, but for all intents and purposes, they're stopped).
RTCPeerConnection.getStats() - Web APIs
exceptions this method does not throw exceptions; instead, it rejects the returned promise with one of the following errors: invalidaccesserror there is no rtcrtpsender or rtcrtpreceiver whose track matches the specified selector, or selector matches more than one sender or receiver.
RTCRemoteOutboundRtpStreamStats.remoteTimestamp - Web APIs
if this property is present, it comes from the rtcp sender report (sr) block, which reflects the clock on the remote peer at the time the message was sent.
RTCRtpCodecCapability - Web APIs
an array of objects of this type is returned in the codecs property of the rtcrtpcapabilities object returned in response to a call to either of the static functions rtcrtpsender.getcapabilities() or rtcrtpreceiver.getcapabilities().
RTCRtpCodecParameters - Web APIs
note: on an rtcrtpreceiver, the format-specific parameters come from the sdp sent by the remote peer, while for rtcrtpsender, they're provided by the local description.
RTCRtpEncodingParameters.scaleResolutionDownBy - Web APIs
therefore, specifying a value less than 1.0 is not permitted and will cause a rangeerror exception to be thrown by rtcpeerconnection.addtransceiver() or rtcrtpsender.setparameters().
RTCRtpReceiver.getCapabilities() static function - Web APIs
you can, similarly, obtain the capabilities of rtcrtpsenders by calling the static function rtcrtpsender.getcapabilities().
RTCRtpSendParameters - Web APIs
the webrtc api's rtcrtpsendparameters dictionary is used to specify the parameters for an rtcrtpsender when calling its setparameters() method.
RTCRtpTransceiver.mid - Web APIs
the read-only rtcrtptransceiver interface's mid property specifies the negotiated media id (mid) which the local and remote peers have agreed upon to uniquely identify the stream's pairing of sender and receiver.
RTCRtpTransceiverInit - Web APIs
sendencodings optional a list of encodings to allow when sending rtp media from the rtcrtpsender.
RTCSessionDescription - Web APIs
the process of negotiating a connection between two peers involves exchanging rtcsessiondescription objects back and forth, with each description suggesting one combination of connection configuration options that the sender of the description supports.
RTCStats - Web APIs
WebAPIRTCStats
rtcstats is the foundation of all webrtc statistics objects rtcrtpstreamstats adds to rtcstats information that applies to all rtp endpoints (that is, both local and remote endpoints, and regardless of whether the endpoint is a sender or a receiver) rtcreceivedrtpstreamstats further adds statistics measured at the receiving end of an rtp stream, regardless of whether it's local or remote.
RTCTrackEvent.receiver - Web APIs
syntax var rtpreceiver = trackevent.receiver; value the rtcrtptransceiver which pairs the receiver with a sender and other properties which establish a single bidirectional srtp stream for use by the track associated with the rtctrackevent.
RTCTrackEventInit.receiver - Web APIs
syntax var trackeventinit = { receiver: rtpreceiver, track: mediastreamtrack, streams: [videostream], transceiver: rtptransceiver }; var rtpreceiver = trackeventinit.receiver; value the rtcrtptransceiver which pairs the receiver with a sender and other properties which establish a single bidirectional srtp stream for use by the track associated with the rtctrackevent.
Keep-Alive - HTTP
the keep-alive general header allows the sender to hint about how the connection may be used to set a timeout and a maximum amount of requests.
Trailer - HTTP
WebHTTPHeadersTrailer
the trailer response header allows the sender to include additional fields at the end of chunked messages in order to supply metadata that might be dynamically generated while the message body is sent, such as a message integrity check, digital signature, or post-processing status.
Via - HTTP
WebHTTPHeadersVia
it is used for tracking message forwards, avoiding request loops, and identifying the protocol capabilities of senders along the request/response chain.