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TCP TURN Integration Test

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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}
This commit is contained in:
Benjamin Wright 2018-04-25 14:06:31 -07:00 committed by Commit Bot
parent 3a9c46d5ff
commit edbd389ecc
1 changed files with 43 additions and 0 deletions
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43
pc/peerconnection_integrationtest.cc
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@ -3887,6 +3887,49 @@ TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
delete SetCalleePcWrapperAndReturnCurrent(nullptr);
}
// Verifies that you can use TCP instead of UDP to connect to a TURN server.
TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
// Enable TCP for the fake turn server.
cricket::TestTurnServer turn_server(
network_thread(), turn_server_internal_address,
turn_server_external_address, cricket::PROTO_TCP);
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
ConnectFakeSignaling();
// Set "offer to receive audio/video" without adding any tracks, so we just
// set up ICE/DTLS with no media.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
EXPECT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
// Need to free the clients here since they're using things we created on
// the stack.
delete SetCallerPcWrapperAndReturnCurrent(nullptr);
delete SetCalleePcWrapperAndReturnCurrent(nullptr);
}
// Test that audio and video flow end-to-end when codec names don't use the
// expected casing, given that they're supposed to be case insensitive. To test
// this, all but one codec is removed from each media description, and its