Fixes crash where packets were sent to a receive stream that had been destroyed but not removed from the ssrc mapping from call to receiver. Added a repro case that reliably crashed before the fix. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2161007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4681 4adac7df-926f-26a2-2b94-8c16560cd09d
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.