Do not regenerate ssrc on SetSendingStatus

webrtc/modules/rtp_rtcp/source/rtp_sender.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 1024 matching lines @@
     return false;
 
   *packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
 
   if (!packet->SetExtension<TransportSequenceNumber>(*packet_id))
     return false;
 
   return true;
 }
 
-void RTPSender::SetSendingStatus(bool enabled) {
-  if (!enabled) {
-    rtc::CritScope lock(&send_critsect_);
-    if (!ssrc_forced_) {
-      // Generate a new SSRC.
-      ssrc_db_->ReturnSSRC(ssrc_);
-      ssrc_ = ssrc_db_->CreateSSRC();
-      RTC_DCHECK(ssrc_ != 0);
-    }
-    // Don't initialize seq number if SSRC passed externally.
-    if (!sequence_number_forced_ && !ssrc_forced_) {
-      // Generate a new sequence number.
-      sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
-    }
-  }
-}
-
 void RTPSender::SetSendingMediaStatus(bool enabled) {
   rtc::CritScope lock(&send_critsect_);
   sending_media_ = enabled;
 }
 
 bool RTPSender::SendingMedia() const {
   rtc::CritScope lock(&send_critsect_);
   return sending_media_;
 }
 
@@ skipping 222 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc