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}
This commit is contained in:
Benjamin Wright 2018-04-26 18:43:40 +00:00 committed by Commit Bot
parent edbd389ecc
commit e782abab19

View File

@ -3887,49 +3887,6 @@ 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