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Make the default value of rtcp-mux policy to required.

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Change the default value of rtcp-mux policy in RTCConfiguration.
Refactor the peerconnectioninterface and webrtcsession unit tests.

BUG=webrtc:6030

Review-Url: https://codereview.webrtc.org/2043193003
Cr-Commit-Position: refs/heads/master@{#15217}
This commit is contained in:
zhihuang 2016-11-23 10:30:12 -08:00 committed by Commit bot
parent e02407ab6c
commit 4dfb8cef51
5 changed files with 35 additions and 12 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/api/android/java/src/org/webrtc/PeerConnection.java
View File

@ -146,7 +146,7 @@ public class PeerConnection {
public RTCConfiguration(List<IceServer> iceServers) {
iceTransportsType = IceTransportsType.ALL;
bundlePolicy = BundlePolicy.BALANCED;
rtcpMuxPolicy = RtcpMuxPolicy.NEGOTIATE;
rtcpMuxPolicy = RtcpMuxPolicy.REQUIRE;
tcpCandidatePolicy = TcpCandidatePolicy.ENABLED;
candidateNetworkPolicy = candidateNetworkPolicy.ALL;
this.iceServers = iceServers;

2
webrtc/api/peerconnectioninterface.h
View File

@ -295,7 +295,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// at the same time.
IceServers servers;
BundlePolicy bundle_policy = kBundlePolicyBalanced;
RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyNegotiate;
RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire;
TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
CandidateNetworkPolicy candidate_network_policy =
kCandidateNetworkPolicyAll;

16
webrtc/api/peerconnectioninterface_unittest.cc
View File

@ -79,6 +79,7 @@ static const char kSdpStringWithStream1[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
@ -86,6 +87,7 @@ static const char kSdpStringWithStream1[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
@ -108,7 +110,8 @@ static const char kSdpStringWithStream1AudioTrackOnly[] =
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n";
"a=ssrc:1 label:audiotrack0\r\n"
"a=rtcp-mux\r\n";
// Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
// MediaStreams have one audio track and one video track.
@ -126,6 +129,7 @@ static const char kSdpStringWithStream1And2[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
@ -134,6 +138,7 @@ static const char kSdpStringWithStream1And2[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
@ -153,10 +158,12 @@ static const char kSdpStringWithoutStreams[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams. Msid is supported.
@ -173,10 +180,12 @@ static const char kSdpStringWithMsidWithoutStreams[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams and audio only.
@ -192,6 +201,7 @@ static const char kSdpStringWithoutStreamsAudioOnly[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
@ -208,11 +218,13 @@ static const char kSdpStringSendOnlyWithoutStreams[] =
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringInit[] =
@ -230,12 +242,14 @@ static const char kSdpStringAudio[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
static const char kSdpStringVideo[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringMs1Audio0[] =

2
webrtc/api/test/testsdpstrings.h
View File

@ -29,6 +29,7 @@ static const char kFireFoxSdpOffer[] =
"c=IN IP4 74.95.2.170\r\n"
"a=rtpmap:109 opus/48000/2\r\n"
"a=ptime:20\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:8 PCMA/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
@ -46,6 +47,7 @@ static const char kFireFoxSdpOffer[] =
" 10.0.254.2 rport 50962\r\n"
"m=video 38826 RTP/SAVPF 120\r\n"
"c=IN IP4 74.95.2.170\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=sendrecv\r\n"
"a=candidate:0 1 UDP 2112946431 172.16.191.1 62017 typ host\r\n"

25
webrtc/api/webrtcsession_unittest.cc
View File

@ -383,7 +383,8 @@ class WebRtcSessionTest
// options. When DTLS is enabled a certificate will be used if provided,
// otherwise one will be generated using the |cert_generator|.
void Init(
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) {
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy) {
ASSERT_TRUE(session_.get() == NULL);
session_.reset(new WebRtcSessionForTest(
media_controller_.get(), rtc::Thread::Current(), rtc::Thread::Current(),
@ -397,6 +398,7 @@ class WebRtcSessionTest
session_->GetOnDestroyedSignal()->connect(
this, &WebRtcSessionTest::OnSessionDestroyed);
configuration_.rtcp_mux_policy = rtcp_mux_policy;
EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
observer_.ice_connection_state_);
EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
@ -415,7 +417,13 @@ class WebRtcSessionTest
void OnSessionDestroyed() { session_destroyed_ = true; }
void Init() { Init(nullptr); }
void Init() {
Init(nullptr, PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
}
void Init(PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy) {
Init(nullptr, rtcp_mux_policy);
}
void InitWithBundlePolicy(
PeerConnectionInterface::BundlePolicy bundle_policy) {
@ -426,8 +434,7 @@ class WebRtcSessionTest
void InitWithRtcpMuxPolicy(
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy) {
PeerConnectionInterface::RTCConfiguration configuration;
configuration_.rtcp_mux_policy = rtcp_mux_policy;
Init();
Init(rtcp_mux_policy);
}
// Successfully init with DTLS; with a certificate generated and supplied or
@ -443,7 +450,8 @@ class WebRtcSessionTest
} else {
RTC_CHECK(false);
}
Init(std::move(cert_generator));
Init(std::move(cert_generator),
PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
}
// Init with DTLS with a store that will fail to generate a certificate.
@ -451,7 +459,8 @@ class WebRtcSessionTest
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
cert_generator->set_should_fail(true);
Init(std::move(cert_generator));
Init(std::move(cert_generator),
PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
}
void InitWithDtmfCodec() {
@ -3347,10 +3356,8 @@ TEST_F(WebRtcSessionTest, TestAddChannelToConnectedBundle) {
// answer to find out if more transports are needed.
configuration_.bundle_policy =
PeerConnectionInterface::kBundlePolicyMaxBundle;
configuration_.rtcp_mux_policy =
PeerConnectionInterface::kRtcpMuxPolicyRequire;
options_.disable_encryption = true;
Init();
Init(PeerConnectionInterface::kRtcpMuxPolicyRequire);
// Negotiate an audio channel with MAX_BUNDLE enabled.
SendAudioOnlyStream2();