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RTC: Refine player and publisher

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pull/2037/head
winlin 4 years ago
parent 3cf3047f97
commit 529264f238
4 changed files with 436 additions and 311 deletions
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324
trunk/research/players/rtc_player.html
Unescape Escape View File

@ -66,8 +66,31 @@
$(function(){
// Async-await-promise based SRS RTC Player.
function SrsRtcPlayerAsync() {
var self = {
play: async function(apiUrl, streamUrl) {
var self = {};
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// or autostart the play:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
// webrtc://r.ossrs.net:11985/myapp/livestream
// or change the stream from livestream to mystream:
// webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream
// or set the candidate(ip) of answer:
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
// or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP:
// webrtc://r.ossrs.net/live/livestream?encrypt=false
// or any other information, will pass-by in the query:
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
// webrtc://r.ossrs.net/live/livestream?token=xxx
self.play = async function(url) {
var conf = self.__internal.prepareUrl(url);
self.pc.addTransceiver("audio", {direction: "recvonly"});
self.pc.addTransceiver("video", {direction: "recvonly"});
@ -76,12 +99,12 @@
var session = await new Promise(function(resolve, reject) {
// @see https://github.com/rtcdn/rtcdn-draft
var data = {
api: apiUrl, streamurl: streamUrl, clientip: null, sdp: offer.sdp
api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
$.ajax({
type: "POST", url: apiUrl, data: JSON.stringify(data),
type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
contentType:'application/json', dataType: 'json'
}).done(function(data) {
console.log("Got answer: ", data);
@ -98,156 +121,100 @@
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
);
return session;
},
close: function() {
};
// Close the publisher.
self.close = function() {
self.pc.close();
},
// callbacks.
onaddstream: function (event) {}
};
self.pc = new RTCPeerConnection(null);
self.pc.onaddstream = function (event) {
if (self.onaddstream) {
self.onaddstream(event);
// The callback when got remote stream.
self.onaddstream = function (event) {};
// Internal APIs.
self.__internal = {
defaultPath: '/rtc/v1/play/',
parse: function (url) {
// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
var a = document.createElement("a");
a.href = url.replace("rtmp://", "http://")
.replace("webrtc://", "http://")
.replace("rtc://", "http://");
var vhost = a.hostname;
var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
// parse the vhost in the params of app, that srs supports.
app = app.replace("...vhost...", "?vhost=");
if (app.indexOf("?") >= 0) {
var params = app.substr(app.indexOf("?"));
app = app.substr(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.substr(0, vhost.indexOf("&"));
}
};
return self;
}
// Promise based SRS RTC Player.
function SrsRtcPlayerPromise() {
var self = {
play: function(apiUrl, streamUrl) {
self.pc.addTransceiver("audio", {direction: "recvonly"});
self.pc.addTransceiver("video", {direction: "recvonly"});
return self.pc.createOffer().then(function(offer) {
return self.pc.setLocalDescription(offer).then(function(){ return offer; });
}).then(function(offer) {
return new Promise(function(resolve, reject) {
// @see https://github.com/rtcdn/rtcdn-draft
var data = {
api: apiUrl, streamurl: streamUrl, clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
$.ajax({
type: "POST", url: apiUrl, data: JSON.stringify(data),
contentType:'application/json', dataType: 'json'
}).done(function(data) {
console.log("Got answer: ", data);
if (data.code) {
reject(data); return;
}
resolve(data);
}).fail(function(reason){
reject(reason);
});
});
}).then(function(session) {
return self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
).then(function(){ return session; });
});
},
close: function() {
self.pc.close();
},
// callbacks.
onaddstream: function (event) {}
};
self.pc = new RTCPeerConnection(null);
self.pc.onaddstream = function (event) {
if (self.onaddstream) {
self.onaddstream(event);
// when vhost equals to server, and server is ip,
// the vhost is __defaultVhost__
if (a.hostname === vhost) {
var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
if (re.test(a.hostname)) {
vhost = "__defaultVhost__";
}
};
return self;
}
// Callback based SRS RTC Player.
function SrsRtcPlayerCallbacks() {
var self = {
play: function(apiUrl, streamUrl, success, fail) {
self.pc.addTransceiver("audio", {direction: "recvonly"});
self.pc.addTransceiver("video", {direction: "recvonly"});
self.pc.createOffer(function(offer){
onOffer(offer);
}, function(reason){
fail(reason);
});
var onOffer = function(offer) {
self.pc.setLocalDescription(offer, function(){
onOfferDone(offer);
}, function(reason) {
fail(reason);
});
};
var onOfferDone = function (offer) {
// @see https://github.com/rtcdn/rtcdn-draft
var data = {
api: apiUrl, streamurl: streamUrl, clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
$.ajax({
type: "POST", url: apiUrl, data: JSON.stringify(data),
contentType:'application/json', dataType: 'json'
}).done(function(data) {
console.log("Got answer: ", data);
if (data.code) {
fail(data); return;
// parse the schema
var schema = "rtmp";
if (url.indexOf("://") > 0) {
schema = url.substr(0, url.indexOf("://"));
}
onAnswer(data);
}).fail(function(reason){
fail(reason);
});
};
var port = a.port;
if (!port) {
if (schema === 'http') {
port = 80;
} else if (schema === 'https') {
port = 443;
} else if (schema === 'rtmp') {
port = 1935;
}
}
var onAnswer = function(session) {
var answer = session.sdp;
self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: answer})
).then(function(){
success(session);
}).catch(function(reason) {
fail(reason);
});
};
},
close: function() {
self.pc.close();
},
// callbacks.
onaddstream: function (event) {}
var ret = {
url: url,
schema: schema,
server: a.hostname, port: port,
vhost: vhost, app: app, stream: stream
};
self.pc = new RTCPeerConnection(null);
self.pc.onaddstream = function (event) {
if (self.onaddstream) {
self.onaddstream(event);
self.__internal.fill_query(a.search, ret);
// For webrtc API, we use 443 if page is https, or schema specified it.
if (!ret.port) {
if (schema === 'webrtc' || schema === 'rtc') {
if (ret.user_query.schema === 'https') {
ret.port = 443;
} else if (window.location.href.indexOf('https://') === 0) {
ret.port = 443;
} else {
// For WebRTC, SRS use 1985 as default API port.
ret.port = 1985;
}
}
};
return self;
}
// Build RTC api url.
var prepareUrl = function () {
var apiUrl, streamUrl;
if (true) {
var urlObject = parse_rtmp_url($("#txt_url").val());
return ret;
},
prepareUrl: function (webrtcUrl) {
var urlObject = self.__internal.parse(webrtcUrl);
// If user specifies the schema, use it as API schema.
var schema = urlObject.user_query.schema;
schema = schema? schema + ':' : window.location.protocol;
schema = schema ? schema + ':' : window.location.protocol;
var port = urlObject.port || 1985;
if (schema === 'https:') {
@ -255,7 +222,7 @@
}
// @see https://github.com/rtcdn/rtcdn-draft
var api = urlObject.user_query.play || '/rtc/v1/play/';
var api = urlObject.user_query.play || self.__internal.defaultPath;
if (api.lastIndexOf('/') !== api.length - 1) {
api += '/';
}
@ -267,66 +234,77 @@
}
}
// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
apiUrl = apiUrl.replace(api + '&', api + '?');
var apiUrl = apiUrl.replace(api + '&', api + '?');
streamUrl = urlObject.url;
}
var streamUrl = urlObject.url;
return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
},
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;
}
}
};
// Start play with conf.
var playStream = function (conf) {
self.pc = new RTCPeerConnection(null);
self.pc.onaddstream = function (event) {
if (self.onaddstream) {
self.onaddstream(event);
}
};
return self;
}
var sdk = null; // Global handler to do cleanup when replaying.
var startPlay = function() {
$('#rtc_media_player').show();
// Close PC when user replay.
if (sdk) {
sdk.close();
}
// Use Callback style.
if (true) {
if (true) {
sdk = new SrsRtcPlayerAsync();
} else {
sdk = new SrsRtcPlayerPromise();
}
sdk.onaddstream = function (event) {
console.log('Start play, event: ', event);
$('#rtc_media_player').prop('srcObject', event.stream);
};
sdk.play(conf.apiUrl, conf.streamUrl).then(function(session){
var simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
// For example:
// webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.play(url).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', simulator + '?drop=1&username=' + session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
} else if (false) {
sdk = new SrsRtcPlayerCallbacks();
sdk.onaddstream = function (event) {
console.log('Start play, event: ', event);
$('#rtc_media_player').prop('srcObject', event.stream);
};
sdk.play(conf.apiUrl, conf.streamUrl, function (session) {
var simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', simulator + '?drop=1&username=' + session.sessionid);
}, function (reason) {
sdk.close();
$('#rtc_media_player').hide();
throw reason;
});
}
};
var sdk = null; // Global handler to do cleanup when replaying.
var startPlay = function() {
$('#rtc_media_player').show();
var conf = prepareUrl();
playStream(conf);
};
$('#rtc_media_player').hide();

291
trunk/research/players/rtc_publisher.html
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@ -65,69 +65,56 @@
<script type="text/javascript">
var pc = null; // Global handler to do cleanup when replaying.
$(function(){
var startPublish = function() {
$('#rtc_media_player').show();
var urlObject = parse_rtmp_url($("#txt_url").val());
var schema = window.location.protocol;
// Close PC when user replay.
if (pc) {
pc.close();
}
// Async-awat-prmise based SRS RTC Publisher.
function SrsRtcPublisherAsync() {
var self = {};
pc = new RTCPeerConnection(null);
pc.addTransceiver("audio", {direction: "sendonly"});
pc.addTransceiver("video", {direction: "sendonly"});
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// or autostart the publish:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
// webrtc://r.ossrs.net:11985/myapp/livestream
// or change the stream from livestream to mystream:
// webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream
// or set the candidate(ip) of answer:
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
// or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP:
// webrtc://r.ossrs.net/live/livestream?encrypt=false
// or any other information, will pass-by in the query:
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
// webrtc://r.ossrs.net/live/livestream?token=xxx
self.publish = async function (url) {
var conf = self.__internal.prepareUrl(url);
self.pc.addTransceiver("audio", {direction: "sendonly"});
self.pc.addTransceiver("video", {direction: "sendonly"});
var constraints = {
audio: true, video: {
height: { max: 320 }
}
};
navigator.mediaDevices.getUserMedia(
constraints
).then(function(stream) {
console.log('Got stream with constraints: ', constraints);
$('#rtc_media_player').prop('srcObject', stream);
pc.addStream(stream);
return new Promise(function(resolve, reject) {
pc.createOffer(function(offer){
resolve(offer);
},function(reason){
reject(reason);
});
var stream = await navigator.mediaDevices.getUserMedia(
{audio: true, video: {height: {max: 320 }}}
);
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
stream.getTracks().forEach(function(track) {
self.pc.addTrack(track);
});
}).then(function(offer) {
return pc.setLocalDescription(offer).then(function(){ return offer; });
}).then(function(offer) {
return new Promise(function(resolve, reject) {
var port = urlObject.port || 1985;
// @see https://github.com/rtcdn/rtcdn-draft
var api = urlObject.user_query.publish || '/rtc/v1/publish/';
if (api.lastIndexOf('/') != api.length - 1) {
api += '/';
}
var url = schema + '//' + urlObject.server + ':' + port + api;
for (var key in urlObject.user_query) {
if (key != 'api' && key != 'publish') {
url += '&' + key + '=' + urlObject.user_query[key];
}
}
// Replace /rtc/v1/publish/&k=v to /rtc/v1/publish/?k=v
url = url.replace(api + '&', api + '?');
self.onaddstream && self.onaddstream({stream: stream});
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
var session = await new Promise(function(resolve, reject) {
// @see https://github.com/rtcdn/rtcdn-draft
var data = {
api: url, streamurl: urlObject.url, clientip: null, sdp: offer.sdp
api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
$.ajax({
type: "POST", url: url, data: JSON.stringify(data),
type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
contentType:'application/json', dataType: 'json'
}).done(function(data) {
console.log("Got answer: ", data);
@ -135,24 +122,194 @@
reject(data); return;
}
var simulator = schema + '//' + urlObject.server + ':' + port + '/rtc/v1/nack/';
$('#sessionid').html(data.sessionid);
$('#simulator-drop').attr('href', simulator + '?drop=1&username=' + data.sessionid);
resolve(data.sdp);
resolve(data);
}).fail(function(reason){
reject(reason);
});
});
}).then(function(answer) {
return pc.setRemoteDescription(new RTCSessionDescription({type: 'answer', sdp: answer}));
}).catch(function(reason) {
pc.getLocalStreams().forEach(function(stream){
stream.getTracks().forEach(function(track) {
track.stop();
});
});
pc.close(); $('#rtc_media_player').hide();
throw reason;
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
);
session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
return session;
};
// Close the publisher.
self.close = function() {
self.pc.close();
};
// The callback when got local stream.
self.onaddstream = function (event) {};
// Internal APIs.
self.__internal = {
defaultPath: '/rtc/v1/publish/',
parse: function (url) {
// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
var a = document.createElement("a");
a.href = url.replace("rtmp://", "http://")
.replace("webrtc://", "http://")
.replace("rtc://", "http://");
var vhost = a.hostname;
var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
// parse the vhost in the params of app, that srs supports.
app = app.replace("...vhost...", "?vhost=");
if (app.indexOf("?") >= 0) {
var params = app.substr(app.indexOf("?"));
app = app.substr(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.substr(0, vhost.indexOf("&"));
}
}
}
// when vhost equals to server, and server is ip,
// the vhost is __defaultVhost__
if (a.hostname === vhost) {
var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
if (re.test(a.hostname)) {
vhost = "__defaultVhost__";
}
}
// parse the schema
var schema = "rtmp";
if (url.indexOf("://") > 0) {
schema = url.substr(0, url.indexOf("://"));
}
var port = a.port;
if (!port) {
if (schema === 'http') {
port = 80;
} else if (schema === 'https') {
port = 443;
} else if (schema === 'rtmp') {
port = 1935;
}
}
var ret = {
url: url,
schema: schema,
server: a.hostname, port: port,
vhost: vhost, app: app, stream: stream
};
self.__internal.fill_query(a.search, ret);
// For webrtc API, we use 443 if page is https, or schema specified it.
if (!ret.port) {
if (schema === 'webrtc' || schema === 'rtc') {
if (ret.user_query.schema === 'https') {
ret.port = 443;
} else if (window.location.href.indexOf('https://') === 0) {
ret.port = 443;
} else {
// For WebRTC, SRS use 1985 as default API port.
ret.port = 1985;
}
}
}
return ret;
},
prepareUrl: function (webrtcUrl) {
var urlObject = self.__internal.parse(webrtcUrl);
// If user specifies the schema, use it as API schema.
var schema = urlObject.user_query.schema;
schema = schema ? schema + ':' : window.location.protocol;
var port = urlObject.port || 1985;
if (schema === 'https:') {
port = urlObject.port || 443;
}
// @see https://github.com/rtcdn/rtcdn-draft
var api = urlObject.user_query.play || self.__internal.defaultPath;
if (api.lastIndexOf('/') !== api.length - 1) {
api += '/';
}
apiUrl = schema + '//' + urlObject.server + ':' + port + api;
for (var key in urlObject.user_query) {
if (key !== 'api' && key !== 'play') {
apiUrl += '&' + key + '=' + urlObject.user_query[key];
}
}
// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
var apiUrl = apiUrl.replace(api + '&', api + '?');
var streamUrl = urlObject.url;
return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
},
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);
return self;
}
var sdk = null; // Global handler to do cleanup when republishing.
var startPublish = function() {
$('#rtc_media_player').show();
// Close PC when user replay.
if (sdk) {
sdk.close();
}
sdk = new SrsRtcPublisherAsync();
sdk.onaddstream = function (event) {
console.log('Start publish, event: ', event);
$('#rtc_media_player').prop('srcObject', event.stream);
};
// For example:
// webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.publish(url).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
};

13
trunk/research/players/srs_publisher.html
Unescape Escape View File

@ -24,11 +24,6 @@
<li><a id="nav_srs_player" href="srs_player.html">SRS播放器</a></li>
<li><a id="nav_rtc_player" href="rtc_player.html">RTC播放器</a></li>
<li><a id="nav_rtc_publisher" href="rtc_publisher.html">RTC推流</a></li>
<li class="active"><a id="nav_srs_publisher" href="srs_publisher.html">SRS编码器</a></li>
<li><a id="nav_srs_chat" href="srs_chat.html">SRS会议</a></li>
<li><a id="nav_srs_bwt" href="srs_bwt.html">SRS测网速</a></li>
<li><a id="nav_vlc" href="vlc.html">VLC播放器</a></li>
<li><a id="nav_gb28181" href="srs_gb28181.html">SRS-GB28181</a></li>
</ul>
</div>
</div>
@ -40,8 +35,8 @@
<button type="button" class="close" data-dismiss="alert">×</button>
<strong><span id="txt_log_title">Warning:</span></strong>
<span id="txt_log_msg">
Flash推流已经很少用，建议用<a href="https://obsproject.com/" target="_blank">OBS</a>或<a href="http://ffmpeg.org/" target="_blank">FFMPEG</a>推流，
如果一定要使用Flash推流请点<a id="https_publisher" href="srs_publisher2.html">这里</a>。
Flash推流已经很少用，建议用<a href="rtc_publisher.html">RTC推流</a>，<a href="https://obsproject.com/" target="_blank">OBS</a>或<a href="http://ffmpeg.org/" target="_blank">FFMPEG</a>推流，
如果一定要使用Flash推流请点<a id="https_publisher" href="srs_publisher_flash.html">这里</a>。
</span>
</div>
<hr/>
@ -56,12 +51,12 @@
$(function(){
var l = window.location;
var url = window.location.href;
if (l.hostname !== 'localhost' && l.hostname !== '127.0.0.1' && l.protocol == 'http:') {
if (l.hostname !== 'localhost' && l.hostname !== '127.0.0.1' && l.protocol === 'http:') {
// For flash publisher, must use HTTPS.
url = window.location.href.replace('http:', 'https:');
}
url = url.substr(0, url.lastIndexOf('/')) + '/srs_publisher2.html';
url = url.substr(0, url.lastIndexOf('/')) + '/srs_publisher_flash.html';
$('#https_publisher').attr('href', url);
});
</script>

5
trunk/research/players/srs_publisher2.html → trunk/research/players/srs_publisher_flash.html
Unescape Escape View File

@ -19,11 +19,6 @@
<div class="nav-collapse collapse">
<ul class="nav">
<li><a id="nav_srs_player" href="srs_player.html">SRS播放器</a></li>
<li class="active"><a id="nav_srs_publisher" href="srs_publisher.html">SRS编码器</a></li>
<li><a id="nav_srs_chat" href="srs_chat.html">SRS会议</a></li>
<li><a id="nav_srs_bwt" href="srs_bwt.html">SRS测网速</a></li>
<li><a id="nav_vlc" href="vlc.html">VLC播放器</a></li>
<li><a id="nav_gb28181" href="srs_gb28181.html">SRS-GB28181</a></li>
</ul>
</div>
</div>
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