Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF. BUG=2796 R=juberti@webrtc.org, pthatcher@google.com Review URL: https://webrtc-codereview.appspot.com/13439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -62,6 +62,10 @@ using talk_base::scoped_ptr;
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// RFC4585
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const char kMediaProtocolAvpf[] = "RTP/AVPF";
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// RFC5124
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const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF";
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// This should be replaced by "UDP/TLS/RTP/SAVPF", but we need to support it for
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// now to be compatible with previous Chrome versions.
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const char kMediaProtocolSavpf[] = "RTP/SAVPF";
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const char kMediaProtocolRtpPrefix[] = "RTP/";
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@ -987,17 +991,20 @@ static bool CreateMediaContentAnswer(
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}
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static bool IsMediaProtocolSupported(MediaType type,
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const std::string& protocol) {
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const std::string& protocol,
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bool secure_transport) {
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// Data channels can have a protocol of SCTP or SCTP/DTLS.
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if (type == MEDIA_TYPE_DATA &&
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(protocol == kMediaProtocolSctp ||
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protocol == kMediaProtocolDtlsSctp)) {
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((protocol == kMediaProtocolSctp && !secure_transport)||
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(protocol == kMediaProtocolDtlsSctp && secure_transport))) {
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return true;
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}
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// Since not all applications serialize and deserialize the media protocol,
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// we will have to accept |protocol| to be empty.
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return protocol == kMediaProtocolAvpf || protocol == kMediaProtocolSavpf ||
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protocol.empty();
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return protocol == kMediaProtocolAvpf || protocol.empty() ||
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protocol == kMediaProtocolSavpf ||
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(protocol == kMediaProtocolDtlsSavpf && secure_transport);
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}
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static void SetMediaProtocol(bool secure_transport,
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@ -1337,8 +1344,9 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
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}
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bool rejected = !options.has_audio || audio_content->rejected ||
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!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
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audio_answer->protocol());
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!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
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audio_answer->protocol(),
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audio_transport->secure());
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if (!rejected) {
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AddTransportAnswer(audio_content->name, *(audio_transport.get()),
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answer.get());
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@ -1385,7 +1393,9 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
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return NULL;
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}
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bool rejected = !options.has_video || video_content->rejected ||
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!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO, video_answer->protocol());
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!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
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video_answer->protocol(),
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video_transport->secure());
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if (!rejected) {
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if (!AddTransportAnswer(video_content->name, *(video_transport.get()),
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answer.get())) {
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@ -1438,7 +1448,9 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
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}
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bool rejected = !options.has_data() || data_content->rejected ||
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!IsMediaProtocolSupported(MEDIA_TYPE_DATA, data_answer->protocol());
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!IsMediaProtocolSupported(MEDIA_TYPE_DATA,
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data_answer->protocol(),
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data_transport->secure());
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if (!rejected) {
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data_answer->set_bandwidth(options.data_bandwidth);
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if (!AddTransportAnswer(data_content->name, *(data_transport.get()),
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@ -80,6 +80,8 @@ extern const char kMediaProtocolAvpf[];
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// RFC5124 RTP/SAVPF
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extern const char kMediaProtocolSavpf[];
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extern const char kMediaProtocolDtlsSavpf[];
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extern const char kMediaProtocolRtpPrefix[];
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extern const char kMediaProtocolSctp[];
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@ -1788,6 +1788,64 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoWithAnswerBundle) {
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TestCryptoWithBundle(false);
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}
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// Verifies that creating answer fails if the offer has UDP/TLS/RTP/SAVPF but
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// DTLS is not enabled locally.
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TEST_F(MediaSessionDescriptionFactoryTest,
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TestOfferDtlsSavpfWithoutDtlsFailed) {
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f1_.set_secure(SEC_ENABLED);
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f2_.set_secure(SEC_ENABLED);
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tdf1_.set_secure(SEC_DISABLED);
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tdf2_.set_secure(SEC_DISABLED);
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talk_base::scoped_ptr<SessionDescription> offer(
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f1_.CreateOffer(MediaSessionOptions(), NULL));
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ASSERT_TRUE(offer.get() != NULL);
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ContentInfo* offer_content = offer->GetContentByName("audio");
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ASSERT_TRUE(offer_content != NULL);
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AudioContentDescription* offer_audio_desc =
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static_cast<AudioContentDescription*>(offer_content->description);
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offer_audio_desc->set_protocol(cricket::kMediaProtocolDtlsSavpf);
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talk_base::scoped_ptr<SessionDescription> answer(
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f2_.CreateAnswer(offer.get(), MediaSessionOptions(), NULL));
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ASSERT_TRUE(answer != NULL);
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ContentInfo* answer_content = answer->GetContentByName("audio");
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ASSERT_TRUE(answer_content != NULL);
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ASSERT_TRUE(answer_content->rejected);
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}
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// Offers UDP/TLS/RTP/SAVPF and verifies the answer can be created and contains
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// UDP/TLS/RTP/SAVPF.
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TEST_F(MediaSessionDescriptionFactoryTest, TestOfferDtlsSavpfCreateAnswer) {
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f1_.set_secure(SEC_ENABLED);
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f2_.set_secure(SEC_ENABLED);
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tdf1_.set_secure(SEC_ENABLED);
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tdf2_.set_secure(SEC_ENABLED);
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talk_base::scoped_ptr<SessionDescription> offer(
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f1_.CreateOffer(MediaSessionOptions(), NULL));
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ASSERT_TRUE(offer.get() != NULL);
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ContentInfo* offer_content = offer->GetContentByName("audio");
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ASSERT_TRUE(offer_content != NULL);
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AudioContentDescription* offer_audio_desc =
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static_cast<AudioContentDescription*>(offer_content->description);
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offer_audio_desc->set_protocol(cricket::kMediaProtocolDtlsSavpf);
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talk_base::scoped_ptr<SessionDescription> answer(
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f2_.CreateAnswer(offer.get(), MediaSessionOptions(), NULL));
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ASSERT_TRUE(answer != NULL);
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const ContentInfo* answer_content = answer->GetContentByName("audio");
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ASSERT_TRUE(answer_content != NULL);
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ASSERT_FALSE(answer_content->rejected);
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const AudioContentDescription* answer_audio_desc =
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static_cast<const AudioContentDescription*>(answer_content->description);
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EXPECT_EQ(std::string(cricket::kMediaProtocolDtlsSavpf),
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answer_audio_desc->protocol());
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}
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// Test that we include both SDES and DTLS in the offer, but only include SDES
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// in the answer if DTLS isn't negotiated.
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TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
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