Adapted tests to API change.

chrome/test/data/webrtc/jsep01_call.js

 /**
  * Copyright (c) 2012 The Chromium Authors. All rights reserved.
  * Use of this source code is governed by a BSD-style license that can be
  * found in the LICENSE file.
  */
 
 // Public interface towards the other javascript files, such as
 // message_handling.js. The contract for these functions is described in
 // message_handling.js.
 
@@ skipping 15 matching lines @@
   } else if (parsed_msg.candidate) {
     var candidate = new RTCIceCandidate(parsed_msg);
     peerConnection.addIceCandidate(candidate);
     return;
   }
   addTestFailure("unknown message received");
   return;
 }
 
 function createPeerConnection(stun_server) {
-  servers = {iceServers:[{uri:"stun:" + stun_server}]};
+  servers = {iceServers:[{url:"stun:" + stun_server}]};
   try {
     peerConnection = new webkitRTCPeerConnection(servers, null);
   } catch (exception) {
     failTest('Failed to create peer connection: ' + exception);
   }
   peerConnection.onaddstream = addStreamCallback_;
   peerConnection.onremovestream = removeStreamCallback_;
   peerConnection.onicecandidate = iceCallback_;
   return peerConnection;
 }
@@ skipping 45 matching lines @@
   // This only mean that we have received a valid SDP message with an offer or
   // an answer, it does not mean that audio and video works.
   returnToTest('ok-got-remote-stream');
 }
 
 /** @private */
 function removeStreamCallback_(event) {
   debug('Call ended.');
   document.getElementById("remote-view").src = '';
 }