Wire up transport sequence number / send time callbacks to webrtc via libjingle.

webrtc/config.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/config.h"
@@ skipping 18 matching lines @@
   return ss.str();
 }
 
 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
 const char* RtpExtension::kAbsSendTime =
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
 const char* RtpExtension::kAudioLevel =
     "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
 const char* RtpExtension::kTransportSequenceNumber =
-    "http://www.webrtc.org/experiments/rtp-hdrext/transport-sequence-number";
+    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
 
 bool RtpExtension::IsSupportedForAudio(const std::string& name) {
   return name == webrtc::RtpExtension::kAbsSendTime ||
          name == webrtc::RtpExtension::kAudioLevel ||
          name == webrtc::RtpExtension::kTransportSequenceNumber;
 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& name) {
   return name == webrtc::RtpExtension::kTOffset ||
          name == webrtc::RtpExtension::kAbsSendTime ||
@@ skipping 63 matching lines @@
   }
   ss << ", encoder_specific_settings: ";
   ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
 
   ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
   ss << '}';
   return ss.str();
 }
 
 }  // namespace webrtc