Transport sequence number should be set also for retransmissions.

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 677 matching lines @@
 }
 
 bool RTPSender::StorePackets() const {
   return packet_history_.StorePackets();
 }
 
 int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
   size_t length = IP_PACKET_SIZE;
   uint8_t data_buffer[IP_PACKET_SIZE];
   int64_t capture_time_ms;
+
   if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
                                                data_buffer, &length,
                                                &capture_time_ms)) {
     // Packet not found.
     return 0;
   }
 
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
@@ skipping 207 matching lines @@
 
   int64_t now_ms = clock_->TimeInMilliseconds();
   int64_t diff_ms = now_ms - capture_time_ms;
   UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
                                diff_ms);
   UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
 
   // TODO(sprang): Potentially too much overhead in IsRegistered()?
   bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
                                  kRtpExtensionTransportSequenceNumber) &&
-                             transport_sequence_number_allocator_ &&
-                             !is_retransmit;
+                             transport_sequence_number_allocator_;
+
   PacketOptions options;
   if (using_transport_seq) {
     options.packet_id =
         UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
   }
 
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
     CriticalSectionScoped lock(send_critsect_.get());
     media_has_been_sent_ = true;
@@ skipping 967 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc