682 lines
26 KiB
JavaScript
682 lines
26 KiB
JavaScript
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//
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// Copyright (c) 2013-2021 Winlin
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//
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// SPDX-License-Identifier: MIT
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//
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'use strict';
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function SrsError(name, message) {
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this.name = name;
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this.message = message;
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this.stack = (new Error()).stack;
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}
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SrsError.prototype = Object.create(Error.prototype);
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SrsError.prototype.constructor = SrsError;
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// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
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// Async-awat-prmise based SRS RTC Publisher.
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function SrsRtcPublisherAsync() {
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var self = {};
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// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
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self.constraints = {
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audio: true,
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video: {
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width: {ideal: 320, max: 576}
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}
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};
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// @see https://github.com/rtcdn/rtcdn-draft
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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// or specifies the API port:
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// webrtc://r.ossrs.net:11985/live/livestream
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// or autostart the publish:
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// webrtc://r.ossrs.net/live/livestream?autostart=true
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// or change the app from live to myapp:
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// webrtc://r.ossrs.net:11985/myapp/livestream
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// or change the stream from livestream to mystream:
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(eip) of answer:
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// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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// webrtc://r.ossrs.net/live/livestream?encrypt=false
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// or any other information, will pass-by in the query:
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// webrtc://r.ossrs.net/live/livestream?vhost=xxx
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// webrtc://r.ossrs.net/live/livestream?token=xxx
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self.publish = async function (url) {
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var conf = self.__internal.prepareUrl(url);
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self.pc.addTransceiver("audio", {direction: "sendonly"});
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self.pc.addTransceiver("video", {direction: "sendonly"});
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//self.pc.addTransceiver("video", {direction: "sendonly"});
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//self.pc.addTransceiver("audio", {direction: "sendonly"});
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if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
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throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
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}
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var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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stream.getTracks().forEach(function (track) {
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self.pc.addTrack(track);
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// Notify about local track when stream is ok.
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self.ontrack && self.ontrack({track: track});
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});
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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var session = await new Promise(function (resolve, reject) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
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clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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const xhr = new XMLHttpRequest();
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xhr.onload = function() {
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if (xhr.readyState !== xhr.DONE) return;
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if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
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const data = JSON.parse(xhr.responseText);
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console.log("Got answer: ", data);
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return data.code ? reject(xhr) : resolve(data);
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}
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xhr.open('POST', conf.apiUrl, true);
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xhr.setRequestHeader('Content-type', 'application/json');
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xhr.send(JSON.stringify(data));
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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);
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session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
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return session;
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};
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// Close the publisher.
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self.close = function () {
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self.pc && self.pc.close();
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self.pc = null;
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};
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// The callback when got local stream.
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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self.ontrack = function (event) {
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// Add track to stream of SDK.
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self.stream.addTrack(event.track);
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};
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// Internal APIs.
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self.__internal = {
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defaultPath: '/rtc/v1/publish/',
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prepareUrl: function (webrtcUrl) {
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var urlObject = self.__internal.parse(webrtcUrl);
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// If user specifies the schema, use it as API schema.
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var schema = urlObject.user_query.schema;
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schema = schema ? schema + ':' : window.location.protocol;
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var port = urlObject.port || 1985;
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if (schema === 'https:') {
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port = urlObject.port || 443;
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}
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// @see https://github.com/rtcdn/rtcdn-draft
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var api = urlObject.user_query.play || self.__internal.defaultPath;
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if (api.lastIndexOf('/') !== api.length - 1) {
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api += '/';
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}
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var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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for (var key in urlObject.user_query) {
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if (key !== 'api' && key !== 'play') {
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apiUrl += '&' + key + '=' + urlObject.user_query[key];
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}
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}
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// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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apiUrl = apiUrl.replace(api + '&', api + '?');
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var streamUrl = urlObject.url;
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return {
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apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
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tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
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};
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},
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parse: function (url) {
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// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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var a = document.createElement("a");
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a.href = url.replace("rtmp://", "http://")
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.replace("webrtc://", "http://")
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.replace("rtc://", "http://");
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var vhost = a.hostname;
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var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
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var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
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// parse the vhost in the params of app, that srs supports.
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app = app.replace("...vhost...", "?vhost=");
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if (app.indexOf("?") >= 0) {
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var params = app.slice(app.indexOf("?"));
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app = app.slice(0, app.indexOf("?"));
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if (params.indexOf("vhost=") > 0) {
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vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
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if (vhost.indexOf("&") > 0) {
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vhost = vhost.slice(0, vhost.indexOf("&"));
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}
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}
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}
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// when vhost equals to server, and server is ip,
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// the vhost is __defaultVhost__
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if (a.hostname === vhost) {
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var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
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if (re.test(a.hostname)) {
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vhost = "__defaultVhost__";
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}
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}
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// parse the schema
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var schema = "rtmp";
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if (url.indexOf("://") > 0) {
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schema = url.slice(0, url.indexOf("://"));
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}
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var port = a.port;
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if (!port) {
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// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
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if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
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port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
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}
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// Guess by schema.
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if (schema === 'http') {
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port = 80;
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} else if (schema === 'https') {
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port = 443;
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} else if (schema === 'rtmp') {
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port = 1935;
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}
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}
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var ret = {
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url: url,
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schema: schema,
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server: a.hostname, port: port,
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vhost: vhost, app: app, stream: stream
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};
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self.__internal.fill_query(a.search, ret);
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// For webrtc API, we use 443 if page is https, or schema specified it.
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if (!ret.port) {
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if (schema === 'webrtc' || schema === 'rtc') {
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if (ret.user_query.schema === 'https') {
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ret.port = 443;
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} else if (window.location.href.indexOf('https://') === 0) {
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ret.port = 443;
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} else {
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// For WebRTC, SRS use 1985 as default API port.
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ret.port = 1985;
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}
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}
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}
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return ret;
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},
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fill_query: function (query_string, obj) {
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// pure user query object.
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obj.user_query = {};
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if (query_string.length === 0) {
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return;
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}
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// split again for angularjs.
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if (query_string.indexOf("?") >= 0) {
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query_string = query_string.split("?")[1];
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}
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var queries = query_string.split("&");
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for (var i = 0; i < queries.length; i++) {
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var elem = queries[i];
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var query = elem.split("=");
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obj[query[0]] = query[1];
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obj.user_query[query[0]] = query[1];
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}
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// alias domain for vhost.
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if (obj.domain) {
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obj.vhost = obj.domain;
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}
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}
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};
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self.pc = new RTCPeerConnection(null);
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// To keep api consistent between player and publisher.
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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// @see https://webrtc.org/getting-started/media-devices
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self.stream = new MediaStream();
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return self;
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}
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// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
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// Async-await-promise based SRS RTC Player.
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function SrsRtcPlayerAsync() {
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var self = {};
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// @see https://github.com/rtcdn/rtcdn-draft
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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// or specifies the API port:
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// webrtc://r.ossrs.net:11985/live/livestream
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// webrtc://r.ossrs.net:80/live/livestream
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// or autostart the play:
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// webrtc://r.ossrs.net/live/livestream?autostart=true
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// or change the app from live to myapp:
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// webrtc://r.ossrs.net:11985/myapp/livestream
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// or change the stream from livestream to mystream:
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(eip) of answer:
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// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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// webrtc://r.ossrs.net/live/livestream?encrypt=false
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// or any other information, will pass-by in the query:
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// webrtc://r.ossrs.net/live/livestream?vhost=xxx
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// webrtc://r.ossrs.net/live/livestream?token=xxx
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self.play = async function(url) {
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var conf = self.__internal.prepareUrl(url);
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self.pc.addTransceiver("audio", {direction: "recvonly"});
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self.pc.addTransceiver("video", {direction: "recvonly"});
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//self.pc.addTransceiver("video", {direction: "recvonly"});
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//self.pc.addTransceiver("audio", {direction: "recvonly"});
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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var session = await new Promise(function(resolve, reject) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
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clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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const xhr = new XMLHttpRequest();
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xhr.onload = function() {
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if (xhr.readyState !== xhr.DONE) return;
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if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
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const data = JSON.parse(xhr.responseText);
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console.log("Got answer: ", data);
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return data.code ? reject(xhr) : resolve(data);
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}
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xhr.open('POST', conf.apiUrl, true);
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xhr.setRequestHeader('Content-type', 'application/json');
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xhr.send(JSON.stringify(data));
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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);
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session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
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return session;
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};
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// Close the player.
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self.close = function() {
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self.pc && self.pc.close();
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self.pc = null;
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};
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// The callback when got remote track.
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// Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
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self.ontrack = function (event) {
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// https://webrtc.org/getting-started/remote-streams
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self.stream.addTrack(event.track);
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};
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// Internal APIs.
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self.__internal = {
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defaultPath: '/rtc/v1/play/',
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prepareUrl: function (webrtcUrl) {
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var urlObject = self.__internal.parse(webrtcUrl);
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// If user specifies the schema, use it as API schema.
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var schema = urlObject.user_query.schema;
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schema = schema ? schema + ':' : window.location.protocol;
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var port = urlObject.port || 1985;
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if (schema === 'https:') {
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port = urlObject.port || 443;
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}
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// @see https://github.com/rtcdn/rtcdn-draft
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var api = urlObject.user_query.play || self.__internal.defaultPath;
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if (api.lastIndexOf('/') !== api.length - 1) {
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api += '/';
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}
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var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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for (var key in urlObject.user_query) {
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if (key !== 'api' && key !== 'play') {
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apiUrl += '&' + key + '=' + urlObject.user_query[key];
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}
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}
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// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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apiUrl = apiUrl.replace(api + '&', api + '?');
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var streamUrl = urlObject.url;
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return {
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apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
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tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
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};
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},
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parse: function (url) {
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// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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var a = document.createElement("a");
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a.href = url.replace("rtmp://", "http://")
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.replace("webrtc://", "http://")
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.replace("rtc://", "http://");
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var vhost = a.hostname;
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var app = a.pathname.substring(1, a.pathname.lastIndexOf("/"));
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var stream = a.pathname.slice(a.pathname.lastIndexOf("/") + 1);
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// parse the vhost in the params of app, that srs supports.
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app = app.replace("...vhost...", "?vhost=");
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if (app.indexOf("?") >= 0) {
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var params = app.slice(app.indexOf("?"));
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app = app.slice(0, app.indexOf("?"));
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if (params.indexOf("vhost=") > 0) {
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vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
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if (vhost.indexOf("&") > 0) {
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vhost = vhost.slice(0, vhost.indexOf("&"));
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}
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}
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}
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// when vhost equals to server, and server is ip,
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// the vhost is __defaultVhost__
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if (a.hostname === vhost) {
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var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
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if (re.test(a.hostname)) {
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vhost = "__defaultVhost__";
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}
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}
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// parse the schema
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var schema = "rtmp";
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if (url.indexOf("://") > 0) {
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schema = url.slice(0, url.indexOf("://"));
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}
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var port = a.port;
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if (!port) {
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// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
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if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
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port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
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}
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// Guess by schema.
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if (schema === 'http') {
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port = 80;
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} else if (schema === 'https') {
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port = 443;
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} else if (schema === 'rtmp') {
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port = 1935;
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}
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}
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var ret = {
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url: url,
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schema: schema,
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server: a.hostname, port: port,
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vhost: vhost, app: app, stream: stream
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};
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self.__internal.fill_query(a.search, ret);
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// For webrtc API, we use 443 if page is https, or schema specified it.
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if (!ret.port) {
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if (schema === 'webrtc' || schema === 'rtc') {
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if (ret.user_query.schema === 'https') {
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ret.port = 443;
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} else if (window.location.href.indexOf('https://') === 0) {
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ret.port = 443;
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} else {
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// For WebRTC, SRS use 1985 as default API port.
|
|
ret.port = 1985;
|
|
}
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
},
|
|
fill_query: function (query_string, obj) {
|
|
// pure user query object.
|
|
obj.user_query = {};
|
|
|
|
if (query_string.length === 0) {
|
|
return;
|
|
}
|
|
|
|
// split again for angularjs.
|
|
if (query_string.indexOf("?") >= 0) {
|
|
query_string = query_string.split("?")[1];
|
|
}
|
|
|
|
var queries = query_string.split("&");
|
|
for (var i = 0; i < queries.length; i++) {
|
|
var elem = queries[i];
|
|
|
|
var query = elem.split("=");
|
|
obj[query[0]] = query[1];
|
|
obj.user_query[query[0]] = query[1];
|
|
}
|
|
|
|
// alias domain for vhost.
|
|
if (obj.domain) {
|
|
obj.vhost = obj.domain;
|
|
}
|
|
}
|
|
};
|
|
|
|
self.pc = new RTCPeerConnection(null);
|
|
|
|
// Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
|
|
self.stream = new MediaStream();
|
|
|
|
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
|
|
self.pc.ontrack = function(event) {
|
|
if (self.ontrack) {
|
|
self.ontrack(event);
|
|
}
|
|
};
|
|
|
|
return self;
|
|
}
|
|
|
|
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
|
|
// Async-awat-prmise based SRS RTC Publisher by WHIP.
|
|
function SrsRtcWhipWhepAsync() {
|
|
var self = {};
|
|
|
|
// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
|
|
self.constraints = {
|
|
audio: true,
|
|
video: {
|
|
width: {ideal: 320, max: 576}
|
|
}
|
|
};
|
|
|
|
// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
|
|
// @url The WebRTC url to publish with, for example:
|
|
// http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
|
|
self.publish = async function (url) {
|
|
if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
|
|
|
|
self.pc.addTransceiver("audio", {direction: "sendonly"});
|
|
self.pc.addTransceiver("video", {direction: "sendonly"});
|
|
|
|
if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
|
|
throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
|
|
}
|
|
var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
|
|
|
|
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
|
|
stream.getTracks().forEach(function (track) {
|
|
self.pc.addTrack(track);
|
|
|
|
// Notify about local track when stream is ok.
|
|
self.ontrack && self.ontrack({track: track});
|
|
});
|
|
|
|
var offer = await self.pc.createOffer();
|
|
await self.pc.setLocalDescription(offer);
|
|
const answer = await new Promise(function (resolve, reject) {
|
|
console.log("Generated offer: ", offer);
|
|
|
|
const xhr = new XMLHttpRequest();
|
|
xhr.onload = function() {
|
|
if (xhr.readyState !== xhr.DONE) return;
|
|
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
|
|
const data = xhr.responseText;
|
|
console.log("Got answer: ", data);
|
|
return data.code ? reject(xhr) : resolve(data);
|
|
}
|
|
xhr.open('POST', url, true);
|
|
xhr.setRequestHeader('Content-type', 'application/sdp');
|
|
xhr.send(offer.sdp);
|
|
});
|
|
await self.pc.setRemoteDescription(
|
|
new RTCSessionDescription({type: 'answer', sdp: answer})
|
|
);
|
|
|
|
return self.__internal.parseId(url, offer.sdp, answer);
|
|
};
|
|
|
|
// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
|
|
// @url The WebRTC url to play with, for example:
|
|
// http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
|
|
self.play = async function(url) {
|
|
if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
|
|
|
|
self.pc.addTransceiver("audio", {direction: "recvonly"});
|
|
self.pc.addTransceiver("video", {direction: "recvonly"});
|
|
|
|
var offer = await self.pc.createOffer();
|
|
await self.pc.setLocalDescription(offer);
|
|
const answer = await new Promise(function(resolve, reject) {
|
|
console.log("Generated offer: ", offer);
|
|
|
|
const xhr = new XMLHttpRequest();
|
|
xhr.onload = function() {
|
|
if (xhr.readyState !== xhr.DONE) return;
|
|
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
|
|
const data = xhr.responseText;
|
|
console.log("Got answer: ", data);
|
|
return data.code ? reject(xhr) : resolve(data);
|
|
}
|
|
xhr.open('POST', url, true);
|
|
xhr.setRequestHeader('Content-type', 'application/sdp');
|
|
xhr.send(offer.sdp);
|
|
});
|
|
await self.pc.setRemoteDescription(
|
|
new RTCSessionDescription({type: 'answer', sdp: answer})
|
|
);
|
|
|
|
return self.__internal.parseId(url, offer.sdp, answer);
|
|
};
|
|
|
|
// Close the publisher.
|
|
self.close = function () {
|
|
self.pc && self.pc.close();
|
|
self.pc = null;
|
|
};
|
|
|
|
// The callback when got local stream.
|
|
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
|
|
self.ontrack = function (event) {
|
|
// Add track to stream of SDK.
|
|
self.stream.addTrack(event.track);
|
|
};
|
|
|
|
self.pc = new RTCPeerConnection(null);
|
|
|
|
// To keep api consistent between player and publisher.
|
|
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
|
|
// @see https://webrtc.org/getting-started/media-devices
|
|
self.stream = new MediaStream();
|
|
|
|
// Internal APIs.
|
|
self.__internal = {
|
|
parseId: (url, offer, answer) => {
|
|
let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
|
|
sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
|
|
sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
|
|
sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
|
|
|
|
const a = document.createElement("a");
|
|
a.href = url;
|
|
return {
|
|
sessionid: sessionid, // Should be ice-ufrag of answer:offer.
|
|
simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
|
|
};
|
|
},
|
|
};
|
|
|
|
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
|
|
self.pc.ontrack = function(event) {
|
|
if (self.ontrack) {
|
|
self.ontrack(event);
|
|
}
|
|
};
|
|
|
|
return self;
|
|
}
|
|
|
|
// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
|
|
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
|
|
function SrsRtcFormatSenders(senders, kind) {
|
|
var codecs = [];
|
|
senders.forEach(function (sender) {
|
|
var params = sender.getParameters();
|
|
params && params.codecs && params.codecs.forEach(function(c) {
|
|
if (kind && sender.track.kind !== kind) {
|
|
return;
|
|
}
|
|
|
|
if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
|
|
return;
|
|
}
|
|
|
|
var s = '';
|
|
|
|
s += c.mimeType.replace('audio/', '').replace('video/', '');
|
|
s += ', ' + c.clockRate + 'HZ';
|
|
if (sender.track.kind === "audio") {
|
|
s += ', channels: ' + c.channels;
|
|
}
|
|
s += ', pt: ' + c.payloadType;
|
|
|
|
codecs.push(s);
|
|
});
|
|
});
|
|
return codecs.join(", ");
|
|
}
|
|
|