Add a configuration parameter for using the media transport for data channels.
Adds a field |use_media_transport_for_data_channels| to RTCConfiguration.
PeerConnection requires a media transport factory to be set if this bit
is set. As with |use_media_transport|, the value may not be modified
after setting the local or remote description.
If either |use_media_transport| or |use_media_transport_for_data_channel| is
set, PeerConnection uses its media transport factory when creating a JSEP
transport controller.
PeerConnection stops unconditionally using media transport in
CreateVoiceChannel, as it may be present only for use in data channels. It uses
the media transport if it is present and |use_media_transport| is set.
Bug: webrtc:9719
Change-Id: I59d4ce8f7531fd19d9c17eefe033f063f663ebcc
Reviewed-on: https://webrtc-review.googlesource.com/c/109041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25507}diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 6504f53..80c3091 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -583,6 +583,15 @@
// provided.
bool use_media_transport = false;
+ // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
+ // informs PeerConnection that it should use the MediaTransportInterface for
+ // data channels. It's invalid to set it to |true| if the
+ // MediaTransportFactory wasn't provided. Data channels over media
+ // transport are not compatible with RTP or SCTP data channels. Setting
+ // both |use_media_transport_for_data_channels| and
+ // |enable_rtp_data_channel| is invalid.
+ bool use_media_transport_for_data_channels = false;
+
// Defines advanced optional cryptographic settings related to SRTP and
// frame encryption for native WebRTC. Setting this will overwrite any
// settings set in PeerConnectionFactory (which is deprecated).
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 0876644..3c5a025 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -706,6 +706,7 @@
absl::optional<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
bool use_media_transport;
+ bool use_media_transport_for_data_channels;
absl::optional<CryptoOptions> crypto_options;
};
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
@@ -756,6 +757,8 @@
network_preference == o.network_preference &&
active_reset_srtp_params == o.active_reset_srtp_params &&
use_media_transport == o.use_media_transport &&
+ use_media_transport_for_data_channels ==
+ o.use_media_transport_for_data_channels &&
crypto_options == o.crypto_options;
}
@@ -946,11 +949,13 @@
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
- if (configuration.use_media_transport) {
+ if (configuration.use_media_transport ||
+ configuration.use_media_transport_for_data_channels) {
if (!factory_->media_transport_factory()) {
RTC_DCHECK(false)
- << "PeerConnecton is initialized with use_media_transport = true, "
- << "but media transport factory is not set in PeerConnectioFactory";
+ << "PeerConnecton is initialized with use_media_transport = true or "
+ << "use_media_transport_for_data_channels = true "
+ << "but media transport factory is not set in PeerConnectionFactory";
return false;
}
@@ -2920,6 +2925,22 @@
}
if (local_description() &&
+ configuration.use_media_transport_for_data_channels !=
+ configuration_.use_media_transport_for_data_channels) {
+ RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels "
+ "after calling SetLocalDescription.";
+ return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
+ }
+
+ if (remote_description() &&
+ configuration.use_media_transport_for_data_channels !=
+ configuration_.use_media_transport_for_data_channels) {
+ RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels "
+ "after calling SetRemoteDescription.";
+ return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
+ }
+
+ if (local_description() &&
configuration.crypto_options != configuration_.crypto_options) {
RTC_LOG(LS_ERROR) << "Can't change crypto_options after calling "
"SetLocalDescription.";
@@ -2951,6 +2972,8 @@
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
modified_config.use_media_transport = configuration.use_media_transport;
+ modified_config.use_media_transport_for_data_channels =
+ configuration.use_media_transport_for_data_channels;
if (configuration != modified_config) {
RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
@@ -3009,8 +3032,10 @@
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
transport_controller_->SetMediaTransportFactory(
- modified_config.use_media_transport ? factory_->media_transport_factory()
- : nullptr);
+ modified_config.use_media_transport ||
+ modified_config.use_media_transport_for_data_channels
+ ? factory_->media_transport_factory()
+ : nullptr);
if (configuration_.active_reset_srtp_params !=
modified_config.active_reset_srtp_params) {
@@ -5597,7 +5622,10 @@
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
const std::string& mid) {
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
- MediaTransportInterface* media_transport = GetMediaTransport(mid);
+ MediaTransportInterface* media_transport = nullptr;
+ if (configuration_.use_media_transport) {
+ media_transport = GetMediaTransport(mid);
+ }
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
call_.get(), configuration_.media_config, rtp_transport, media_transport,
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index e725798..4099992 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -938,10 +938,13 @@
// to use media transport. Otherwise returns nullptr.
MediaTransportInterface* GetMediaTransport(const std::string& mid) {
auto media_transport = transport_controller_->GetMediaTransport(mid);
- RTC_DCHECK(configuration_.use_media_transport ==
+ RTC_DCHECK((configuration_.use_media_transport ||
+ configuration_.use_media_transport_for_data_channels) ==
(media_transport != nullptr))
<< "configuration_.use_media_transport="
<< configuration_.use_media_transport
+ << ", configuration_.use_media_transport_for_data_channels="
+ << configuration_.use_media_transport_for_data_channels
<< ", (media_transport != nullptr)=" << (media_transport != nullptr);
return media_transport;
}
diff --git a/pc/peerconnection_media_unittest.cc b/pc/peerconnection_media_unittest.cc
index e01ebb8..3c069ae 100644
--- a/pc/peerconnection_media_unittest.cc
+++ b/pc/peerconnection_media_unittest.cc
@@ -1129,6 +1129,74 @@
ASSERT_EQ(nullptr, callee_video->media_transport());
}
+TEST_P(PeerConnectionMediaTest, MediaTransportOnlyForDataChannels) {
+ RTCConfiguration config;
+
+ // Setup PeerConnection to use media transport for data channels.
+ config.use_media_transport_for_data_channels = true;
+
+ // Force SDES.
+ config.enable_dtls_srtp = false;
+
+ auto caller = CreatePeerConnectionWithAudioVideo(config);
+ auto callee = CreatePeerConnectionWithAudioVideo(config);
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
+
+ auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
+ auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
+ ASSERT_TRUE(caller_voice);
+ ASSERT_TRUE(callee_voice);
+
+ // Make sure media transport is not propagated to voice channel.
+ EXPECT_EQ(nullptr, caller_voice->media_transport());
+ EXPECT_EQ(nullptr, callee_voice->media_transport());
+}
+
+TEST_P(PeerConnectionMediaTest, MediaTransportForMediaAndDataChannels) {
+ RTCConfiguration config;
+
+ // Setup PeerConnection to use media transport for both media and data
+ // channels.
+ config.use_media_transport = true;
+ config.use_media_transport_for_data_channels = true;
+
+ // Force SDES.
+ config.enable_dtls_srtp = false;
+
+ auto caller = CreatePeerConnectionWithAudioVideo(config);
+ auto callee = CreatePeerConnectionWithAudioVideo(config);
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
+
+ auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
+ auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
+ ASSERT_TRUE(caller_voice);
+ ASSERT_TRUE(callee_voice);
+
+ // Make sure media transport is propagated to voice channel.
+ FakeMediaTransport* caller_voice_media_transport =
+ static_cast<FakeMediaTransport*>(caller_voice->media_transport());
+ FakeMediaTransport* callee_voice_media_transport =
+ static_cast<FakeMediaTransport*>(callee_voice->media_transport());
+ ASSERT_NE(nullptr, caller_voice_media_transport);
+ ASSERT_NE(nullptr, callee_voice_media_transport);
+
+ // Make sure media transport is created with correct is_caller.
+ EXPECT_TRUE(caller_voice_media_transport->is_caller());
+ EXPECT_FALSE(callee_voice_media_transport->is_caller());
+
+ // TODO(sukhanov): Propagate media transport to video channel. This test
+ // will fail once media transport is propagated to video channel and it will
+ // serve as a reminder to add a test for video channel propagation.
+ auto caller_video = caller->media_engine()->GetVideoChannel(0);
+ auto callee_video = callee->media_engine()->GetVideoChannel(0);
+ ASSERT_EQ(nullptr, caller_video->media_transport());
+ ASSERT_EQ(nullptr, callee_video->media_transport());
+}
+
TEST_P(PeerConnectionMediaTest, MediaTransportNotPropagatedToVoiceEngine) {
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
diff --git a/pc/peerconnectioninterface_unittest.cc b/pc/peerconnectioninterface_unittest.cc
index 55932b6..12d192e 100644
--- a/pc/peerconnectioninterface_unittest.cc
+++ b/pc/peerconnectioninterface_unittest.cc
@@ -1431,12 +1431,14 @@
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.type = PeerConnectionInterface::kRelay;
config.use_media_transport = true;
+ config.use_media_transport_for_data_channels = true;
EXPECT_TRUE(pc_->SetConfiguration(config));
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
EXPECT_TRUE(returned_config.use_media_transport);
+ EXPECT_TRUE(returned_config.use_media_transport_for_data_channels);
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
diff --git a/sdk/android/api/org/webrtc/PeerConnection.java b/sdk/android/api/org/webrtc/PeerConnection.java
index 1cb5717..dac32c8 100644
--- a/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/sdk/android/api/org/webrtc/PeerConnection.java
@@ -468,6 +468,12 @@
*/
public boolean useMediaTransport;
+ /*
+ * Experimental flag that enables a use of media transport for data channels. If this is true,
+ * the media transport factory MUST be provided to the PeerConnectionFactory.
+ */
+ public boolean useMediaTransportForDataChannels;
+
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
@@ -515,6 +521,7 @@
sdpSemantics = SdpSemantics.PLAN_B;
activeResetSrtpParams = false;
useMediaTransport = false;
+ useMediaTransportForDataChannels = false;
cryptoOptions = null;
}
@@ -720,6 +727,11 @@
return useMediaTransport;
}
+ @CalledByNative("RTCConfiguration")
+ boolean getUseMediaTransportForDataChannels() {
+ return useMediaTransportForDataChannels;
+ }
+
@Nullable
@CalledByNative("RTCConfiguration")
CryptoOptions getCryptoOptions() {
diff --git a/sdk/android/src/jni/pc/peerconnection.cc b/sdk/android/src/jni/pc/peerconnection.cc
index 0d51847..162f809 100644
--- a/sdk/android/src/jni/pc/peerconnection.cc
+++ b/sdk/android/src/jni/pc/peerconnection.cc
@@ -249,6 +249,9 @@
Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
rtc_config->use_media_transport =
Java_RTCConfiguration_getUseMediaTransport(jni, j_rtc_config);
+ rtc_config->use_media_transport_for_data_channels =
+ Java_RTCConfiguration_getUseMediaTransportForDataChannels(jni,
+ j_rtc_config);
rtc_config->crypto_options =
JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
}
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.h b/sdk/objc/api/peerconnection/RTCConfiguration.h
index fcaa07a..f140081 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -189,6 +189,12 @@
@property(nonatomic, assign) BOOL useMediaTransport;
/**
+ * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
+ * that it should use the MediaTransportInterface for data channels.
+ */
+@property(nonatomic, assign) BOOL useMediaTransportForDataChannels;
+
+/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
* options set through the PeerConnectionFactory (which is deprecated).
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 83f32b4..55bf26f 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -51,6 +51,7 @@
@synthesize turnCustomizer = _turnCustomizer;
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
@synthesize useMediaTransport = _useMediaTransport;
+@synthesize useMediaTransportForDataChannels = _useMediaTransportForDataChannels;
@synthesize cryptoOptions = _cryptoOptions;
- (instancetype)init {
@@ -100,6 +101,7 @@
_iceBackupCandidatePairPingInterval =
config.ice_backup_candidate_pair_ping_interval;
_useMediaTransport = config.use_media_transport;
+ _useMediaTransportForDataChannels = config.use_media_transport_for_data_channels;
_keyType = RTCEncryptionKeyTypeECDSA;
_iceCandidatePoolSize = config.ice_candidate_pool_size;
_shouldPruneTurnPorts = config.prune_turn_ports;
@@ -199,6 +201,7 @@
nativeConfig->ice_backup_candidate_pair_ping_interval =
_iceBackupCandidatePairPingInterval;
nativeConfig->use_media_transport = _useMediaTransport;
+ nativeConfig->use_media_transport_for_data_channels = _useMediaTransportForDataChannels;
rtc::KeyType keyType =
[[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
if (_certificate != nullptr) {
