Making candidate pool size behave as decided in JSEP.

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 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 1262 matching lines @@
     if (rtc::starts_with(data_desc->protocol().data(),
                          cricket::kMediaProtocolRtpPrefix)) {
       UpdateLocalRtpDataChannels(data_desc->streams());
     }
   }
 
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
   signaling_thread()->Post(RTC_FROM_HERE, this,
                            MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
 
+  // According to JSEP, after setLocalDescription, changing the candidate pool
+  // size is not allowed, and changing the set of ICE servers will not result
+  // in new candidates being gathered.
+  port_allocator_->FreezeCandidatePool();
+
   // MaybeStartGathering needs to be called after posting
   // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
   // before signaling that SetLocalDescription completed.
   session_->MaybeStartGathering();
+
+  if (desc->type() == SessionDescriptionInterface::kAnswer) {
+    // TODO(deadbeef): We already had to hop to the network thread for
+    // MaybeStartGathering...
+    network_thread()->Invoke<void>(
+        RTC_FROM_HERE,
+        rtc::Bind(&cricket::PortAllocator::DiscardPooledSessions,
+                  port_allocator_.get()));
+  }
 }
 
 void PeerConnection::SetRemoteDescription(
     SetSessionDescriptionObserver* observer,
     SessionDescriptionInterface* desc) {
   TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
   if (IsClosed()) {
     return;
   }
   if (!observer) {
@@ skipping 90 matching lines @@
     observer_->OnAddStream(new_stream);
     observer_->OnAddStream(
         rtc::scoped_refptr<MediaStreamInterface>(new_stream));
   }
 
   UpdateEndedRemoteMediaStreams();
 
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
   signaling_thread()->Post(RTC_FROM_HERE, this,
                            MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
+
+  if (desc->type() == SessionDescriptionInterface::kAnswer) {
+    // TODO(deadbeef): We already had to hop to the network thread for
+    // MaybeStartGathering...
+    network_thread()->Invoke<void>(
+        RTC_FROM_HERE,
+        rtc::Bind(&cricket::PortAllocator::DiscardPooledSessions,
+                  port_allocator_.get()));
+  }
 }
 
 PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
   return configuration_;
 }
 
 bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
                                       RTCError* error) {
   TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
 
@@ skipping 145 matching lines @@
 }
 
 void PeerConnection::Close() {
   TRACE_EVENT0("webrtc", "PeerConnection::Close");
   // Update stats here so that we have the most recent stats for tracks and
   // streams before the channels are closed.
   stats_->UpdateStats(kStatsOutputLevelStandard);
 
   session_->Close();
   event_log_.reset();
+  network_thread()->Invoke<void>(
+      RTC_FROM_HERE,
+      rtc::Bind(&cricket::PortAllocator::DiscardPooledSessions,
+                port_allocator_.get()));
 }
 
 void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
                                           WebRtcSession::State state) {
   switch (state) {
     case WebRtcSession::STATE_INIT:
       ChangeSignalingState(PeerConnectionInterface::kStable);
       break;
     case WebRtcSession::STATE_SENTOFFER:
       ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
@@ skipping 1007 matching lines @@
   }
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   if (event_log_) {
     event_log_->StopLogging();
   }
 }
 }  // namespace webrtc