Wire up RTP keep-alive in ortc api.
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
diff --git a/webrtc/ortc/ortcrtpsender_unittest.cc b/webrtc/ortc/ortcrtpsender_unittest.cc
index ab8d821..a94ed76 100644
--- a/webrtc/ortc/ortcrtpsender_unittest.cc
+++ b/webrtc/ortc/ortcrtpsender_unittest.cc
@@ -40,10 +40,10 @@
nullptr, nullptr, nullptr, nullptr, nullptr,
std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_));
ortc_factory_ = ortc_factory_result.MoveValue();
- RtcpParameters rtcp_parameters;
- rtcp_parameters.mux = true;
+ RtpTransportParameters parameters;
+ parameters.rtcp.mux = true;
auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
- rtcp_parameters, &fake_packet_transport_, nullptr, nullptr);
+ parameters, &fake_packet_transport_, nullptr, nullptr);
rtp_transport_ = rtp_transport_result.MoveValue();
}
@@ -153,10 +153,10 @@
// test/tests for it.
TEST_F(OrtcRtpSenderTest, SetTransportFails) {
rtc::FakePacketTransport fake_packet_transport("another_transport");
- RtcpParameters rtcp_parameters;
- rtcp_parameters.mux = true;
+ RtpTransportParameters parameters;
+ parameters.rtcp.mux = true;
auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
- rtcp_parameters, &fake_packet_transport, nullptr, nullptr);
+ parameters, &fake_packet_transport, nullptr, nullptr);
auto rtp_transport = rtp_transport_result.MoveValue();
auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,