Update the call when the network route changes

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 71 matching lines @@
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
+  void OnNetworkRouteChanged(
+      const std::string& transport_name,
+      const cricket::NetworkRoute& network_route) override;
+
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
                         int64_t rtt_ms) override;
 
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
-
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   VoiceEngine* voice_engine() {
     internal::AudioState* audio_state =
         static_cast<internal::AudioState*>(config_.audio_state.get());
     if (audio_state)
       return audio_state->voice_engine();
     else
       return nullptr;
@@ skipping 47 matching lines @@
   int64_t last_rtp_packet_received_ms_;
   int64_t first_packet_sent_ms_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
 
+  std::map<std::string, cricket::NetworkRoute> network_routes_;
+
   VieRemb remb_;
   const std::unique_ptr<CongestionController> congestion_controller_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
@@ skipping 401 matching lines @@
     ReadLockScoped read_lock(*receive_crit_);
     for (auto& kv : audio_receive_ssrcs_) {
       kv.second->SignalNetworkState(audio_network_state_);
     }
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(video_network_state_);
     }
   }
 }
 
+// TODO(honghaiz): Add tests for this method.
+void Call::OnNetworkRouteChanged(const std::string& transport_name,
+                                 const cricket::NetworkRoute& network_route) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  // Check if the network route is connected.
+  if (!network_route.connected) {
+    LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
+    // TODO(honghaiz): Perhaps it should stop sending until a connected route is
+    // established.

[stefan-webrtc, 2016/03/31 07:17:05]

This makes me wonder if this method should replace SignalChannelNetworkState
too? Or should we differentiate between a NetworkRoute which isn't .connected
and one which is "down" (as set by SignalChannelNetworkState)?

[honghaiz3, 2016/03/31 17:42:52]

Currently SignalChannelNetworkState is triggered by
transport_channel::writable() and EWOULDBLOCK, but this is signalled by selected
candidate pair changed. 
So I think this should not replace SignalChannelNetworkState. Perhaps this
(network disconnect) need not be handled here. Instead, it should be handled in
the SignalChannelNetworkState (network up and down).
It is also possible to merge them together, but this will need some more
discussion. Peter and I discussed the plan to refactor the SignalReadyToSend
(which fires SignalChannelNetworkState). We can consider merging these two
methods along with that work. For now I will just leave a TODO here.

[stefan-webrtc, 2016/04/01 08:08:38]

Sounds good, it would be nice to keep the number of "network has changed in some
way" signals down...

[honghaiz3, 2016/04/01 22:21:07]

Not sure what you mean by that. 

+    return;
+  }
+
+  // Check whether the network route has changed on each transport.
+  auto result =
+      network_routes_.insert(std::make_pair(transport_name, network_route));
+  auto kv = result.first;
+  bool inserted = result.second;
+  if (inserted) {
+    // No need to reset BWE if this is the first time the network connects.
+    return;
+  }
+  bool changed = kv->second != network_route;
+  if (changed) {
+    kv->second = network_route;
+    LOG(LS_INFO) << "Network route changed on transport " << transport_name
+                 << ": new local network id " << network_route.local_network_id
+                 << " new remote network id "
+                 << network_route.remote_network_id;
+    // TODO(stefan@): Update the BWE bitrates.

[stefan-webrtc, 2016/03/31 07:17:05]

TODO(holmer) preferred

[honghaiz3, 2016/03/31 17:42:52]

Done.

+  }
+}
+
 void Call::UpdateAggregateNetworkState() {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   bool have_audio = false;
   bool have_video = false;
   {
     ReadLockScoped read_lock(*send_crit_);
     if (audio_send_ssrcs_.size() > 0)
       have_audio = true;
     if (video_send_ssrcs_.size() > 0)
@@ skipping 194 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc