Migrating Apprtc to use new TURN service which supports time-limited TURN credentials.
Review URL: https://webrtc-codereview.appspot.com/1291004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3773 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/samples/js/apprtc/apprtc.py b/samples/js/apprtc/apprtc.py
index 4de7a05..0547970 100644
--- a/samples/js/apprtc/apprtc.py
+++ b/samples/js/apprtc/apprtc.py
@@ -315,6 +315,7 @@
min_re = self.request.get('minre')
max_re = self.request.get('maxre')
hd_video = self.request.get('hd')
+ turn_url = 'https://computeengineondemand.appspot.com/'
if hd_video.lower() == 'true':
min_re = '1280x720'
ts_pwd = self.request.get('tp')
@@ -374,6 +375,7 @@
room_link = base_url + '?r=' + room_key
room_link = append_url_arguments(self.request, room_link)
+ turn_url = turn_url + 'turn?' + 'username=' + user + '&key=4080218913'
token = create_channel(room, user, token_timeout)
pc_config = make_pc_config(stun_server, turn_server, ts_pwd)
pc_constraints = make_pc_constraints(compat)
@@ -387,7 +389,8 @@
'pc_config': json.dumps(pc_config),
'pc_constraints': json.dumps(pc_constraints),
'offer_constraints': json.dumps(offer_constraints),
- 'media_constraints': json.dumps(media_constraints)
+ 'media_constraints': json.dumps(media_constraints),
+ 'turn_url': turn_url
}
if unittest:
target_page = 'test/test_' + unittest + '.html'
diff --git a/samples/js/apprtc/index.html b/samples/js/apprtc/index.html
index ebf35be..e87bfd5 100644
--- a/samples/js/apprtc/index.html
+++ b/samples/js/apprtc/index.html
@@ -112,10 +112,13 @@
var remoteStream;
var channel;
var channelReady = false;
+ var turnReady = false;
var pc;
var socket;
var initiator = {{ initiator }};
var started = false;
+ var pc_config = {{ pc_config|safe }};
+ var pc_constraints = {{ pc_constraints|safe }};
// Set up audio and video regardless of what devices are present.
var sdpConstraints = {'mandatory': {
'OfferToReceiveAudio':true,
@@ -133,6 +136,7 @@
// NOTE: AppRTCClient.java searches & parses this line; update there when
// changing here.
openChannel('{{ token }}');
+ requestTurn('{{ turn_url }}');
doGetUserMedia();
}
@@ -148,6 +152,35 @@
socket = channel.open(handler);
}
+ function requestTurn(turn_url) {
+ var turnExists = false;
+ for (var i in pc_config.iceServers) {
+ if (pc_config.iceServers[i].url.substr(0, 5) == 'turn:') {
+ turnExists = true;
+ turnReady = true;
+ break;
+ }
+ }
+ if (!turnExists) {
+ // No turn server. Get one from computeengineondemand.appspot.com:
+ xmlhttp = new XMLHttpRequest();
+ xmlhttp.onreadystatechange = onTurnResult;
+ xmlhttp.open("GET", turn_url, true);
+ xmlhttp.send();
+ }
+ }
+
+ function onTurnResult() {
+ if (xmlhttp.readyState == 4 && xmlhttp.status == 200) {
+ var turnServer = JSON.parse(xmlhttp.responseText);
+ pc_config.iceServers.push({
+ "url": "turn:" + turnServer.username + "@" + turnServer.turn,
+ "credential": turnServer.password
+ });
+ turnReady = true;
+ }
+ }
+
function resetStatus() {
if (!initiator) {
setStatus("Waiting for someone to join: <a href=\"{{ room_link }}\">{{ room_link }}</a>");
@@ -170,13 +203,11 @@
}
}
- function createPeerConnection() {
- var pc_config = {{ pc_config|safe }};
- var pc_constraints = {{ pc_constraints|safe }};
+ function createPeerConnection() {
// Force the use of a number IP STUN server for Firefox.
if (webrtcDetectedBrowser == "firefox") {
pc_config = {"iceServers":[{"url":"stun:23.21.150.121"}]};
- }
+ }
try {
// Create an RTCPeerConnection via the polyfill (adapter.js).
pc = new RTCPeerConnection(pc_config, pc_constraints);
@@ -189,13 +220,12 @@
alert("Cannot create RTCPeerConnection object; WebRTC is not supported by this browser.");
return;
}
-
pc.onaddstream = onRemoteStreamAdded;
pc.onremovestream = onRemoteStreamRemoved;
}
function maybeStart() {
- if (!started && localStream && channelReady) {
+ if (!started && localStream && channelReady && turnReady) {
setStatus("Connecting...");
console.log("Creating PeerConnection.");
createPeerConnection();