Revert "TCP TURN Integration Test"
This reverts commit edbd389ecc7388973b55e6e3787ed6a95254dc99. Reason for revert: Breaking integration on Linux. Original change's description: > TCP TURN Integration Test > > This changeset adds a new integration test to do basic validation that TCP > TURN functionality works in WebRTC. It simply sets up a TestTurnServer > configured to relay over TCP and then allows the clients to connect to this > server over TCP. > > Bug: webrtc:7668 > Change-Id: Id9f3b4e22f40ace7c7eeddf103b5d954a0872777 > Reviewed-on: https://webrtc-review.googlesource.com/70568 > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23044} TBR=deadbeef@webrtc.org,stefan@webrtc.org,pthatcher@webrtc.org,benwright@webrtc.org Change-Id: Icdf8747d7a1a7bd2a1a29f1536821a0eacb7764e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7668 Reviewed-on: https://webrtc-review.googlesource.com/72961 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23045}
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@ -3887,49 +3887,6 @@ TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
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delete SetCalleePcWrapperAndReturnCurrent(nullptr);
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}
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// Verifies that you can use TCP instead of UDP to connect to a TURN server.
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TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
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static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
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3478};
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static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
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// Enable TCP for the fake turn server.
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cricket::TestTurnServer turn_server(
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network_thread(), turn_server_internal_address,
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turn_server_external_address, cricket::PROTO_TCP);
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webrtc::PeerConnectionInterface::IceServer ice_server;
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ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
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ice_server.username = "test";
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ice_server.password = "test";
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PeerConnectionInterface::RTCConfiguration client_1_config;
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client_1_config.servers.push_back(ice_server);
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client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
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PeerConnectionInterface::RTCConfiguration client_2_config;
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client_2_config.servers.push_back(ice_server);
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client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
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ASSERT_TRUE(
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CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
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ConnectFakeSignaling();
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// Set "offer to receive audio/video" without adding any tracks, so we just
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// set up ICE/DTLS with no media.
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PeerConnectionInterface::RTCOfferAnswerOptions options;
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options.offer_to_receive_audio = 1;
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options.offer_to_receive_video = 1;
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caller()->SetOfferAnswerOptions(options);
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caller()->CreateAndSetAndSignalOffer();
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EXPECT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
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// Need to free the clients here since they're using things we created on
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// the stack.
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delete SetCallerPcWrapperAndReturnCurrent(nullptr);
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delete SetCalleePcWrapperAndReturnCurrent(nullptr);
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}
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// Test that audio and video flow end-to-end when codec names don't use the
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// expected casing, given that they're supposed to be case insensitive. To test
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// this, all but one codec is removed from each media description, and its
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