From f65de8483e90d1d52d5d8f40f646e77bf45b10ea Mon Sep 17 00:00:00 2001
From: Donald Curtis <decurtis@webrtc.org>
Date: Fri, 22 May 2015 14:55:19 -0700
Subject: [PATCH] Fix sending wrong candidates down to transportchannel.

BUG=4665
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54489004

Cr-Commit-Position: refs/heads/master@{#9266}
---
 talk/app/webrtc/webrtcsession_unittest.cc | 55 +++++++++++++++++++++++
 webrtc/p2p/base/p2ptransportchannel.h     |  3 ++
 webrtc/p2p/base/session.cc                |  7 +++
 3 files changed, 65 insertions(+)

diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 2b6f207e2..f21319b15 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -40,10 +40,13 @@
 #include "talk/media/base/fakevideorenderer.h"
 #include "talk/media/base/mediachannel.h"
 #include "talk/media/devices/fakedevicemanager.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/dtlstransportchannel.h"
 #include "webrtc/p2p/base/stunserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannelproxy.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
@@ -2602,6 +2605,58 @@ TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionInvalidIceCredentials) {
   EXPECT_FALSE(session_->SetRemoteDescription(modified_offer, &error));
 }
 
+// Test that candidates sent to the "video" transport do not get pushed down to
+// the "audio" transport channel when bundling using TransportProxy.
+TEST_F(WebRtcSessionTest, TestBalancedBundleCandidateMisuse) {
+  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
+
+  InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  options.use_rtp_mux = true;
+
+  SessionDescriptionInterface* offer = CreateRemoteOffer();
+  SetRemoteDescriptionWithoutError(offer);
+
+  SessionDescriptionInterface* answer = CreateAnswer(NULL);
+  SetLocalDescriptionWithoutError(answer);
+
+  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
+            session_->GetTransportProxy("video")->impl());
+
+  cricket::Candidate candidate0;
+  candidate0.set_address(rtc::SocketAddress("1.1.1.1", 5000));
+  candidate0.set_component(1);
+  candidate0.set_protocol("udp");
+  JsepIceCandidate ice_candidate0(kMediaContentName0, kMediaContentIndex0,
+                                  candidate0);
+  EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate0));
+  // Let the candidate get processed completely.
+  rtc::Thread::Current()->ProcessMessages(100);
+
+  cricket::Transport* t = session_->GetTransport("audio");
+  cricket::TransportStats stats;
+
+  t->GetStats(&stats);
+  EXPECT_EQ(1, stats.channel_stats.size());
+  EXPECT_EQ(2, stats.channel_stats[0].connection_infos.size());
+
+  cricket::Candidate candidate1;
+  candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5001));
+  candidate1.set_component(1);
+  candidate1.set_protocol("udp");
+  JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1,
+                                  candidate1);
+  EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
+  // Let the candidate get processed completely.
+  rtc::Thread::Current()->ProcessMessages(100);
+
+  t->GetStats(&stats);
+  EXPECT_EQ(1, stats.channel_stats.size());
+  EXPECT_EQ(2, stats.channel_stats[0].connection_infos.size());
+}
+
 // kBundlePolicyBalanced bundle policy and answer contains BUNDLE.
 TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
diff --git a/webrtc/p2p/base/p2ptransportchannel.h b/webrtc/p2p/base/p2ptransportchannel.h
index a014701f1..ab1d53c86 100644
--- a/webrtc/p2p/base/p2ptransportchannel.h
+++ b/webrtc/p2p/base/p2ptransportchannel.h
@@ -156,6 +156,9 @@ class P2PTransportChannel : public TransportChannelImpl,
   // Public for unit tests.
   Connection* FindNextPingableConnection();
 
+  // Public for unit tests only.
+  size_t RemoteCandidateCount() const { return remote_candidates_.size(); }
+
  private:
   rtc::Thread* thread() { return worker_thread_; }
   PortAllocatorSession* allocator_session() {
diff --git a/webrtc/p2p/base/session.cc b/webrtc/p2p/base/session.cc
index 38c983689..8cbe128a7 100644
--- a/webrtc/p2p/base/session.cc
+++ b/webrtc/p2p/base/session.cc
@@ -259,6 +259,13 @@ bool TransportProxy::OnRemoteCandidates(const Candidates& candidates,
   // TODO(juberti): Remove this once everybody calls SetLocalTD.
   CompleteNegotiation();
 
+  // Ignore candidates for if the proxy content_name doesn't match the content
+  // name of the actual transport. This stops video candidates from being sent
+  // down to the audio transport when BUNDLE is enabled.
+  if (content_name_ != transport_->get()->content_name()) {
+    return true;
+  }
+
   // Verify each candidate before passing down to transport layer.
   for (Candidates::const_iterator cand = candidates.begin();
        cand != candidates.end(); ++cand) {