Do not regenerate ssrc on SetSendingStatus

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  Copyright (c) 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 337 matching lines @@
   BitrateSent(&state.send_bitrate, &tmp, &tmp, &tmp);
   return state;
 }
 
 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false
     if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
       LOG(LS_WARNING) << "Failed to send RTCP BYE";
     }
-
-    collision_detected_ = false;
-
-    // Generate a new SSRC for the next "call" if false
-    rtp_sender_.SetSendingStatus(sending);
-
-    // Make sure that RTCP objects are aware of our SSRC (it could have changed
-    // Due to collision)
-    uint32_t SSRC = rtp_sender_.SSRC();
-    rtcp_sender_.SetSSRC(SSRC);
-    SetRtcpReceiverSsrcs(SSRC);
-
-    return 0;
   }
   return 0;
 }
 
 bool ModuleRtpRtcpImpl::Sending() const {
   return rtcp_sender_.Sending();
 }
 
 void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
   rtp_sender_.SetSendingMediaStatus(sending);
@@ skipping 530 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc