Only generate one CNAME per PeerConnection.
The CNAME is generated in the PeerConnection constructor and is populated through the MediaSessionOptions.
A default cname will be set in the MediaSessionOptions constructor.
BUG=webrtc:3431
Review-Url: https://codereview.webrtc.org/1871993002
Cr-Commit-Position: refs/heads/master@{#12650}
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index 7f1f452..506a215 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -66,6 +66,9 @@
// NOTE: Must be in the same order as the ServiceType enum.
static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
+// The length of RTCP CNAMEs.
+static const int kRtcpCnameLength = 16;
+
// NOTE: A loop below assumes that the first value of this enum is 0 and all
// other values are incremental.
enum ServiceType {
@@ -377,6 +380,16 @@
namespace webrtc {
+// Generate a RTCP CNAME when a PeerConnection is created.
+std::string GenerateRtcpCname() {
+ std::string cname;
+ if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
+ LOG(LS_ERROR) << "Failed to generate CNAME.";
+ RTC_DCHECK(false);
+ }
+ return cname;
+}
+
bool ExtractMediaSessionOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
bool is_offer,
@@ -508,6 +521,7 @@
ice_state_(kIceNew),
ice_connection_state_(kIceConnectionNew),
ice_gathering_state_(kIceGatheringNew),
+ rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()) {}
@@ -1503,6 +1517,8 @@
if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
+
+ session_options->rtcp_cname = rtcp_cname_;
return true;
}
@@ -1540,6 +1556,8 @@
if (!ParseConstraintsForAnswer(constraints, session_options)) {
return false;
}
+ session_options->rtcp_cname = rtcp_cname_;
+
FinishOptionsForAnswer(session_options);
return true;
}
@@ -1552,6 +1570,8 @@
if (!ExtractMediaSessionOptions(options, false, session_options)) {
return false;
}
+ session_options->rtcp_cname = rtcp_cname_;
+
FinishOptionsForAnswer(session_options);
return true;
}
diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h
index b557715..862c6fb 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/api/peerconnection.h
@@ -369,6 +369,10 @@
std::unique_ptr<cricket::PortAllocator> port_allocator_;
std::unique_ptr<MediaControllerInterface> media_controller_;
+ // One PeerConnection has only one RTCP CNAME.
+ // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
+ std::string rtcp_cname_;
+
// Streams added via AddStream.
rtc::scoped_refptr<StreamCollection> local_streams_;
// Streams created as a result of SetRemoteDescription.
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 1a8dd57..2594b6c 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -934,6 +934,34 @@
ASSERT_TRUE(stream->AddTrack(video_track));
}
+ rtc::scoped_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
+ CreatePeerConnection();
+ AddVoiceStream(kStreamLabel1);
+ rtc::scoped_ptr<SessionDescriptionInterface> offer;
+ EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
+ return offer;
+ }
+
+ rtc::scoped_ptr<SessionDescriptionInterface>
+ CreateAnswerWithOneAudioStream() {
+ rtc::scoped_ptr<SessionDescriptionInterface> offer =
+ CreateOfferWithOneAudioStream();
+ EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
+ rtc::scoped_ptr<SessionDescriptionInterface> answer;
+ EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
+ return answer;
+ }
+
+ const std::string& GetFirstAudioStreamCname(
+ const SessionDescriptionInterface* desc) {
+ const cricket::ContentInfo* audio_content =
+ cricket::GetFirstAudioContent(desc->description());
+ const cricket::AudioContentDescription* audio_desc =
+ static_cast<const cricket::AudioContentDescription*>(
+ audio_content->description);
+ return audio_desc->streams()[0].cname;
+ }
+
cricket::FakePortAllocator* port_allocator_ = nullptr;
scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
scoped_refptr<PeerConnectionInterface> pc_;
@@ -941,6 +969,27 @@
rtc::scoped_refptr<StreamCollection> reference_collection_;
};
+// Generate different CNAMEs when PeerConnections are created.
+// The CNAMEs are expected to be generated randomly. It is possible
+// that the test fails, though the possibility is very low.
+TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
+ rtc::scoped_ptr<SessionDescriptionInterface> offer1 =
+ CreateOfferWithOneAudioStream();
+ rtc::scoped_ptr<SessionDescriptionInterface> offer2 =
+ CreateOfferWithOneAudioStream();
+ EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
+ GetFirstAudioStreamCname(offer2.get()));
+}
+
+TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
+ rtc::scoped_ptr<SessionDescriptionInterface> answer1 =
+ CreateAnswerWithOneAudioStream();
+ rtc::scoped_ptr<SessionDescriptionInterface> answer2 =
+ CreateAnswerWithOneAudioStream();
+ EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
+ GetFirstAudioStreamCname(answer2.get()));
+}
+
TEST_F(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentConfigurations) {
CreatePeerConnectionWithDifferentConfigurations();