09e8c47ee5
r4374 was mistakenly committed to stable, so this is to re-merge back to trunk. Store the sequence number in StopSend() and resume it in StartSend(). When restarting the microphone device, we call StopSend() first, then StartSend() later. Since we reset sequence number in StopSend(), it sometimes causes libSRTP to complain about packets being replayed. Libjingle work around it by caching the sequence number in WebRtcVoiceEngine.cc, and call SetInitSequenceNumber() to resume the sequence number before StartSend().Store the sequence number in StopSend() and resume it in StartSend(). When restarting the microphone device, we call StopSend() first, then StartSend() later. Since we reset sequence number in StopSend(), it sometimes causes libSRTP to complain about packets being replayed. Libjingle work around it by caching the sequence number in WebRtcVoiceEngine.cc, and call SetInitSequenceNumber() to resume the sequence number before StartSend(). This patch fixes this problem by storing the sequence number in StopSend(), and resume it in StartSend(). So that we can remove the workaround in libjingle. BUG=2102 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1922004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4451 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.