Only generate one CNAME per PeerConnection.

webrtc/api/peerconnection.h

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 350 matching lines @@
   UMAObserver* uma_observer_;
   SignalingState signaling_state_;
   // TODO(bemasc): Remove ice_state_.
   IceState ice_state_;
   IceConnectionState ice_connection_state_;
   IceGatheringState ice_gathering_state_;
 
   rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
   rtc::scoped_ptr<MediaControllerInterface> media_controller_;
 
+  // One PeerConnection has only one CNAME.

[pthatcher1, 2016/05/05 18:50:51]

Can you add a link to where this is specified?  I think it's here:  

https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9

[Zhi Huang, 2016/05/06 01:36:53]

Done.

+  std::string cname_;

[pthatcher1, 2016/05/05 18:50:51]

Can you call this rtcp_cname_?  

And "RTCP CNAME" in the comment?

[Zhi Huang, 2016/05/06 01:36:53]

Done.

+
   // Streams added via AddStream.
   rtc::scoped_refptr<StreamCollection> local_streams_;
   // Streams created as a result of SetRemoteDescription.
   rtc::scoped_refptr<StreamCollection> remote_streams_;
 
   std::vector<rtc::scoped_ptr<MediaStreamObserver>> stream_observers_;
 
   // These lists store track info seen in local/remote descriptions.
   TrackInfos remote_audio_tracks_;
   TrackInfos remote_video_tracks_;
@@ skipping 15 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTION_H_