| /* |
| * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "api/peer_connection_interface.h" |
| |
| #include <limits.h> |
| #include <stdint.h> |
| #include <string.h> |
| |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/strings/str_replace.h" |
| #include "absl/types/optional.h" |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio_codecs/audio_decoder_factory.h" |
| #include "api/audio_codecs/audio_encoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/call/call_factory_interface.h" |
| #include "api/create_peerconnection_factory.h" |
| #include "api/data_channel_interface.h" |
| #include "api/jsep.h" |
| #include "api/jsep_session_description.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_types.h" |
| #include "api/rtc_error.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/rtc_event_log/rtc_event_log_factory.h" |
| #include "api/rtc_event_log_output.h" |
| #include "api/rtc_event_log_output_file.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "api/rtp_sender_interface.h" |
| #include "api/rtp_transceiver_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/transport/field_trial_based_config.h" |
| #include "api/video_codecs/builtin_video_decoder_factory.h" |
| #include "api/video_codecs/builtin_video_encoder_factory.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "media/base/codec.h" |
| #include "media/base/media_config.h" |
| #include "media/base/media_engine.h" |
| #include "media/base/stream_params.h" |
| #include "media/engine/webrtc_media_engine.h" |
| #include "media/engine/webrtc_media_engine_defaults.h" |
| #include "media/sctp/sctp_transport_internal.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "p2p/base/fake_port_allocator.h" |
| #include "p2p/base/p2p_constants.h" |
| #include "p2p/base/port.h" |
| #include "p2p/base/port_allocator.h" |
| #include "p2p/base/transport_description.h" |
| #include "p2p/base/transport_info.h" |
| #include "pc/audio_track.h" |
| #include "pc/media_session.h" |
| #include "pc/media_stream.h" |
| #include "pc/peer_connection.h" |
| #include "pc/peer_connection_factory.h" |
| #include "pc/rtc_stats_collector.h" |
| #include "pc/rtp_sender.h" |
| #include "pc/session_description.h" |
| #include "pc/stream_collection.h" |
| #include "pc/test/fake_audio_capture_module.h" |
| #include "pc/test/fake_rtc_certificate_generator.h" |
| #include "pc/test/fake_video_track_source.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "pc/test/test_sdp_strings.h" |
| #include "pc/video_track.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/socket_address.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/time_utils.h" |
| #include "rtc_base/virtual_socket_server.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/file_utils.h" |
| |
| #ifdef WEBRTC_ANDROID |
| #include "pc/test/android_test_initializer.h" |
| #endif |
| |
| namespace webrtc { |
| namespace { |
| |
| static const char kStreamId1[] = "local_stream_1"; |
| static const char kStreamId2[] = "local_stream_2"; |
| static const char kStreamId3[] = "local_stream_3"; |
| static const int kDefaultStunPort = 3478; |
| static const char kStunAddressOnly[] = "stun:address"; |
| static const char kStunInvalidPort[] = "stun:address:-1"; |
| static const char kStunAddressPortAndMore1[] = "stun:address:port:more"; |
| static const char kStunAddressPortAndMore2[] = "stun:address:port more"; |
| static const char kTurnIceServerUri[] = "turn:turn.example.org"; |
| static const char kTurnUsername[] = "user"; |
| static const char kTurnPassword[] = "password"; |
| static const char kTurnHostname[] = "turn.example.org"; |
| static const uint32_t kTimeout = 10000U; |
| |
| static const char kStreams[][8] = {"stream1", "stream2"}; |
| static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"}; |
| static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"}; |
| |
| static const char kRecvonly[] = "recvonly"; |
| static const char kSendrecv[] = "sendrecv"; |
| |
| // Reference SDP with a MediaStream with label "stream1" and audio track with |
| // id "audio_1" and a video track with id "video_1; |
| static const char kSdpStringWithStream1PlanB[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 mslabel:stream1\r\n" |
| "a=ssrc:1 label:audiotrack0\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:video\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/90000\r\n" |
| "a=ssrc:2 cname:stream1\r\n" |
| "a=ssrc:2 mslabel:stream1\r\n" |
| "a=ssrc:2 label:videotrack0\r\n"; |
| // Same string as above but with the MID changed to the Unified Plan default. |
| // This is needed so that this SDP can be used as an answer for a Unified Plan |
| // offer. |
| static const char kSdpStringWithStream1UnifiedPlan[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:0\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 mslabel:stream1\r\n" |
| "a=ssrc:1 label:audiotrack0\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:1\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/90000\r\n" |
| "a=ssrc:2 cname:stream1\r\n" |
| "a=ssrc:2 mslabel:stream1\r\n" |
| "a=ssrc:2 label:videotrack0\r\n"; |
| |
| // Reference SDP with a MediaStream with label "stream1" and audio track with |
| // id "audio_1"; |
| static const char kSdpStringWithStream1AudioTrackOnly[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 mslabel:stream1\r\n" |
| "a=ssrc:1 label:audiotrack0\r\n" |
| "a=rtcp-mux\r\n"; |
| |
| // Reference SDP with two MediaStreams with label "stream1" and "stream2. Each |
| // MediaStreams have one audio track and one video track. |
| // This uses MSID. |
| static const char kSdpStringWithStream1And2PlanB[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=msid-semantic: WMS stream1 stream2\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 msid:stream1 audiotrack0\r\n" |
| "a=ssrc:3 cname:stream2\r\n" |
| "a=ssrc:3 msid:stream2 audiotrack1\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:video\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/0\r\n" |
| "a=ssrc:2 cname:stream1\r\n" |
| "a=ssrc:2 msid:stream1 videotrack0\r\n" |
| "a=ssrc:4 cname:stream2\r\n" |
| "a=ssrc:4 msid:stream2 videotrack1\r\n"; |
| static const char kSdpStringWithStream1And2UnifiedPlan[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=msid-semantic: WMS stream1 stream2\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:0\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 msid:stream1 audiotrack0\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:1\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/0\r\n" |
| "a=ssrc:2 cname:stream1\r\n" |
| "a=ssrc:2 msid:stream1 videotrack0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:2\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "a=ssrc:3 cname:stream2\r\n" |
| "a=ssrc:3 msid:stream2 audiotrack1\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:3\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/0\r\n" |
| "a=ssrc:4 cname:stream2\r\n" |
| "a=ssrc:4 msid:stream2 videotrack1\r\n"; |
| |
| // Reference SDP without MediaStreams. Msid is not supported. |
| static const char kSdpStringWithoutStreams[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:video\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/90000\r\n"; |
| |
| // Reference SDP without MediaStreams. Msid is supported. |
| static const char kSdpStringWithMsidWithoutStreams[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=msid-semantic: WMS\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:video\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/90000\r\n"; |
| |
| // Reference SDP without MediaStreams and audio only. |
| static const char kSdpStringWithoutStreamsAudioOnly[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n"; |
| |
| // Reference SENDONLY SDP without MediaStreams. Msid is not supported. |
| static const char kSdpStringSendOnlyWithoutStreams[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=sendonly\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n" |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:video\r\n" |
| "a=sendrecv\r\n" |
| "a=sendonly\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/90000\r\n"; |
| |
| static const char kSdpStringInit[] = |
| "v=0\r\n" |
| "o=- 0 0 IN IP4 127.0.0.1\r\n" |
| "s=-\r\n" |
| "t=0 0\r\n" |
| "a=msid-semantic: WMS\r\n"; |
| |
| static const char kSdpStringAudio[] = |
| "m=audio 1 RTP/AVPF 103\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:audio\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:103 ISAC/16000\r\n"; |
| |
| static const char kSdpStringVideo[] = |
| "m=video 1 RTP/AVPF 120\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" |
| "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" |
| "a=mid:video\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=rtpmap:120 VP8/90000\r\n"; |
| |
| static const char kSdpStringMs1Audio0[] = |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 msid:stream1 audiotrack0\r\n"; |
| |
| static const char kSdpStringMs1Video0[] = |
| "a=ssrc:2 cname:stream1\r\n" |
| "a=ssrc:2 msid:stream1 videotrack0\r\n"; |
| |
| static const char kSdpStringMs1Audio1[] = |
| "a=ssrc:3 cname:stream1\r\n" |
| "a=ssrc:3 msid:stream1 audiotrack1\r\n"; |
| |
| static const char kSdpStringMs1Video1[] = |
| "a=ssrc:4 cname:stream1\r\n" |
| "a=ssrc:4 msid:stream1 videotrack1\r\n"; |
| |
| static const char kDtlsSdesFallbackSdp[] = |
| "v=0\r\n" |
| "o=xxxxxx 7 2 IN IP4 0.0.0.0\r\n" |
| "s=-\r\n" |
| "c=IN IP4 0.0.0.0\r\n" |
| "t=0 0\r\n" |
| "a=group:BUNDLE audio\r\n" |
| "a=msid-semantic: WMS\r\n" |
| "m=audio 1 RTP/SAVPF 0\r\n" |
| "a=sendrecv\r\n" |
| "a=rtcp-mux\r\n" |
| "a=mid:audio\r\n" |
| "a=ssrc:1 cname:stream1\r\n" |
| "a=ssrc:1 mslabel:stream1\r\n" |
| "a=ssrc:1 label:audiotrack0\r\n" |
| "a=ice-ufrag:e5785931\r\n" |
| "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" |
| "a=rtpmap:0 pcmu/8000\r\n" |
| "a=fingerprint:sha-1 " |
| "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n" |
| "a=setup:actpass\r\n" |
| "a=crypto:0 AES_CM_128_HMAC_SHA1_80 " |
| "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 " |
| "dummy_session_params\r\n"; |
| |
| class RtcEventLogOutputNull final : public RtcEventLogOutput { |
| public: |
| bool IsActive() const override { return true; } |
| bool Write(const std::string& output) override { return true; } |
| }; |
| |
| using ::cricket::StreamParams; |
| using ::testing::Exactly; |
| using ::testing::Values; |
| |
| using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions; |
| |
| // Gets the first ssrc of given content type from the ContentInfo. |
| bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) { |
| if (!content_info || !ssrc) { |
| return false; |
| } |
| const cricket::MediaContentDescription* media_desc = |
| content_info->media_description(); |
| if (!media_desc || media_desc->streams().empty()) { |
| return false; |
| } |
| *ssrc = media_desc->streams().begin()->first_ssrc(); |
| return true; |
| } |
| |
| // Get the ufrags out of an SDP blob. Useful for testing ICE restart |
| // behavior. |
| std::vector<std::string> GetUfrags( |
| const webrtc::SessionDescriptionInterface* desc) { |
| std::vector<std::string> ufrags; |
| for (const cricket::TransportInfo& info : |
| desc->description()->transport_infos()) { |
| ufrags.push_back(info.description.ice_ufrag); |
| } |
| return ufrags; |
| } |
| |
| void SetSsrcToZero(std::string* sdp) { |
| const char kSdpSsrcAtribute[] = "a=ssrc:"; |
| const char kSdpSsrcAtributeZero[] = "a=ssrc:0"; |
| size_t ssrc_pos = 0; |
| while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) != |
| std::string::npos) { |
| size_t end_ssrc = sdp->find(" ", ssrc_pos); |
| sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero); |
| ssrc_pos = end_ssrc; |
| } |
| } |
| |
| // Check if |streams| contains the specified track. |
| bool ContainsTrack(const std::vector<cricket::StreamParams>& streams, |
| const std::string& stream_id, |
| const std::string& track_id) { |
| for (const cricket::StreamParams& params : streams) { |
| if (params.first_stream_id() == stream_id && params.id == track_id) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // Check if |senders| contains the specified sender, by id. |
| bool ContainsSender( |
| const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders, |
| const std::string& id) { |
| for (const auto& sender : senders) { |
| if (sender->id() == id) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // Check if |senders| contains the specified sender, by id and stream id. |
| bool ContainsSender( |
| const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders, |
| const std::string& id, |
| const std::string& stream_id) { |
| for (const auto& sender : senders) { |
| if (sender->id() == id && sender->stream_ids()[0] == stream_id) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // Create a collection of streams. |
| // CreateStreamCollection(1) creates a collection that |
| // correspond to kSdpStringWithStream1. |
| // CreateStreamCollection(2) correspond to kSdpStringWithStream1And2. |
| rtc::scoped_refptr<StreamCollection> CreateStreamCollection( |
| int number_of_streams, |
| int tracks_per_stream) { |
| rtc::scoped_refptr<StreamCollection> local_collection( |
| StreamCollection::Create()); |
| |
| for (int i = 0; i < number_of_streams; ++i) { |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream( |
| webrtc::MediaStream::Create(kStreams[i])); |
| |
| for (int j = 0; j < tracks_per_stream; ++j) { |
| // Add a local audio track. |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( |
| webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], |
| nullptr)); |
| stream->AddTrack(audio_track); |
| |
| // Add a local video track. |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( |
| webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j], |
| webrtc::FakeVideoTrackSource::Create(), |
| rtc::Thread::Current())); |
| stream->AddTrack(video_track); |
| } |
| |
| local_collection->AddStream(stream); |
| } |
| return local_collection; |
| } |
| |
| // Check equality of StreamCollections. |
| bool CompareStreamCollections(StreamCollectionInterface* s1, |
| StreamCollectionInterface* s2) { |
| if (s1 == nullptr || s2 == nullptr || s1->count() != s2->count()) { |
| return false; |
| } |
| |
| for (size_t i = 0; i != s1->count(); ++i) { |
| if (s1->at(i)->id() != s2->at(i)->id()) { |
| return false; |
| } |
| webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks(); |
| webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks(); |
| webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks(); |
| webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks(); |
| |
| if (audio_tracks1.size() != audio_tracks2.size()) { |
| return false; |
| } |
| for (size_t j = 0; j != audio_tracks1.size(); ++j) { |
| if (audio_tracks1[j]->id() != audio_tracks2[j]->id()) { |
| return false; |
| } |
| } |
| if (video_tracks1.size() != video_tracks2.size()) { |
| return false; |
| } |
| for (size_t j = 0; j != video_tracks1.size(); ++j) { |
| if (video_tracks1[j]->id() != video_tracks2[j]->id()) { |
| return false; |
| } |
| } |
| } |
| return true; |
| } |
| |
| // Helper class to test Observer. |
| class MockTrackObserver : public ObserverInterface { |
| public: |
| explicit MockTrackObserver(NotifierInterface* notifier) |
| : notifier_(notifier) { |
| notifier_->RegisterObserver(this); |
| } |
| |
| ~MockTrackObserver() { Unregister(); } |
| |
| void Unregister() { |
| if (notifier_) { |
| notifier_->UnregisterObserver(this); |
| notifier_ = nullptr; |
| } |
| } |
| |
| MOCK_METHOD(void, OnChanged, (), (override)); |
| |
| private: |
| NotifierInterface* notifier_; |
| }; |
| |
| // The PeerConnectionMediaConfig tests below verify that configuration and |
| // constraints are propagated into the PeerConnection's MediaConfig. These |
| // settings are intended for MediaChannel constructors, but that is not |
| // exercised by these unittest. |
| class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory { |
| public: |
| static rtc::scoped_refptr<PeerConnectionFactoryForTest> |
| CreatePeerConnectionFactoryForTest() { |
| PeerConnectionFactoryDependencies dependencies; |
| dependencies.worker_thread = rtc::Thread::Current(); |
| dependencies.network_thread = rtc::Thread::Current(); |
| dependencies.signaling_thread = rtc::Thread::Current(); |
| dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); |
| dependencies.trials = std::make_unique<FieldTrialBasedConfig>(); |
| cricket::MediaEngineDependencies media_deps; |
| media_deps.task_queue_factory = dependencies.task_queue_factory.get(); |
| // Use fake audio device module since we're only testing the interface |
| // level, and using a real one could make tests flaky when run in parallel. |
| media_deps.adm = FakeAudioCaptureModule::Create(); |
| SetMediaEngineDefaults(&media_deps); |
| media_deps.trials = dependencies.trials.get(); |
| dependencies.media_engine = |
| cricket::CreateMediaEngine(std::move(media_deps)); |
| dependencies.call_factory = webrtc::CreateCallFactory(); |
| dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>( |
| dependencies.task_queue_factory.get()); |
| |
| return new rtc::RefCountedObject<PeerConnectionFactoryForTest>( |
| std::move(dependencies)); |
| } |
| |
| using PeerConnectionFactory::PeerConnectionFactory; |
| |
| private: |
| rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| }; |
| |
| // TODO(steveanton): Convert to use the new PeerConnectionWrapper. |
| class PeerConnectionInterfaceBaseTest : public ::testing::Test { |
| protected: |
| explicit PeerConnectionInterfaceBaseTest(SdpSemantics sdp_semantics) |
| : vss_(new rtc::VirtualSocketServer()), |
| main_(vss_.get()), |
| sdp_semantics_(sdp_semantics) { |
| #ifdef WEBRTC_ANDROID |
| webrtc::InitializeAndroidObjects(); |
| #endif |
| } |
| |
| void SetUp() override { |
| // Use fake audio capture module since we're only testing the interface |
| // level, and using a real one could make tests flaky when run in parallel. |
| fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); |
| pc_factory_ = webrtc::CreatePeerConnectionFactory( |
| rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), |
| rtc::scoped_refptr<webrtc::AudioDeviceModule>( |
| fake_audio_capture_module_), |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| webrtc::CreateBuiltinVideoEncoderFactory(), |
| webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */, |
| nullptr /* audio_processing */); |
| ASSERT_TRUE(pc_factory_); |
| pc_factory_for_test_ = |
| PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest(); |
| } |
| |
| void TearDown() override { |
| if (pc_) |
| pc_->Close(); |
| } |
| |
| void CreatePeerConnection() { |
| CreatePeerConnection(PeerConnectionInterface::RTCConfiguration()); |
| } |
| |
| // DTLS does not work in a loopback call, so is disabled for most of the |
| // tests in this file. |
| void CreatePeerConnectionWithoutDtls() { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = false; |
| |
| CreatePeerConnection(config); |
| } |
| |
| void CreatePeerConnectionWithIceTransportsType( |
| PeerConnectionInterface::IceTransportsType type) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.type = type; |
| return CreatePeerConnection(config); |
| } |
| |
| void CreatePeerConnectionWithIceServer(const std::string& uri, |
| const std::string& username, |
| const std::string& password) { |
| PeerConnectionInterface::RTCConfiguration config; |
| PeerConnectionInterface::IceServer server; |
| server.uri = uri; |
| server.username = username; |
| server.password = password; |
| config.servers.push_back(server); |
| CreatePeerConnection(config); |
| } |
| |
| void CreatePeerConnection(const RTCConfiguration& config) { |
| if (pc_) { |
| pc_->Close(); |
| pc_ = nullptr; |
| } |
| std::unique_ptr<cricket::FakePortAllocator> port_allocator( |
| new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr)); |
| port_allocator_ = port_allocator.get(); |
| |
| // Create certificate generator unless DTLS constraint is explicitly set to |
| // false. |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator; |
| |
| if (config.enable_dtls_srtp.value_or(true)) { |
| fake_certificate_generator_ = new FakeRTCCertificateGenerator(); |
| cert_generator.reset(fake_certificate_generator_); |
| } |
| RTCConfiguration modified_config = config; |
| modified_config.sdp_semantics = sdp_semantics_; |
| pc_ = pc_factory_->CreatePeerConnection( |
| modified_config, std::move(port_allocator), std::move(cert_generator), |
| &observer_); |
| ASSERT_TRUE(pc_.get() != NULL); |
| observer_.SetPeerConnectionInterface(pc_.get()); |
| EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); |
| } |
| |
| void CreatePeerConnectionExpectFail(const std::string& uri) { |
| PeerConnectionInterface::RTCConfiguration config; |
| PeerConnectionInterface::IceServer server; |
| server.uri = uri; |
| config.servers.push_back(server); |
| config.sdp_semantics = sdp_semantics_; |
| rtc::scoped_refptr<PeerConnectionInterface> pc = |
| pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_); |
| EXPECT_EQ(nullptr, pc); |
| } |
| |
| void CreatePeerConnectionExpectFail( |
| PeerConnectionInterface::RTCConfiguration config) { |
| PeerConnectionInterface::IceServer server; |
| server.uri = kTurnIceServerUri; |
| server.password = kTurnPassword; |
| config.servers.push_back(server); |
| config.sdp_semantics = sdp_semantics_; |
| rtc::scoped_refptr<PeerConnectionInterface> pc = |
| pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_); |
| EXPECT_EQ(nullptr, pc); |
| } |
| |
| void CreatePeerConnectionWithDifferentConfigurations() { |
| CreatePeerConnectionWithIceServer(kStunAddressOnly, "", ""); |
| EXPECT_EQ(1u, port_allocator_->stun_servers().size()); |
| EXPECT_EQ(0u, port_allocator_->turn_servers().size()); |
| EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname()); |
| EXPECT_EQ(kDefaultStunPort, |
| port_allocator_->stun_servers().begin()->port()); |
| |
| CreatePeerConnectionExpectFail(kStunInvalidPort); |
| CreatePeerConnectionExpectFail(kStunAddressPortAndMore1); |
| CreatePeerConnectionExpectFail(kStunAddressPortAndMore2); |
| |
| CreatePeerConnectionWithIceServer(kTurnIceServerUri, kTurnUsername, |
| kTurnPassword); |
| EXPECT_EQ(0u, port_allocator_->stun_servers().size()); |
| EXPECT_EQ(1u, port_allocator_->turn_servers().size()); |
| EXPECT_EQ(kTurnUsername, |
| port_allocator_->turn_servers()[0].credentials.username); |
| EXPECT_EQ(kTurnPassword, |
| port_allocator_->turn_servers()[0].credentials.password); |
| EXPECT_EQ(kTurnHostname, |
| port_allocator_->turn_servers()[0].ports[0].address.hostname()); |
| } |
| |
| void ReleasePeerConnection() { |
| pc_ = NULL; |
| observer_.SetPeerConnectionInterface(NULL); |
| } |
| |
| rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack( |
| const std::string& label) { |
| return pc_factory_->CreateVideoTrack(label, FakeVideoTrackSource::Create()); |
| } |
| |
| void AddVideoTrack(const std::string& track_label, |
| const std::vector<std::string>& stream_ids = {}) { |
| auto sender_or_error = |
| pc_->AddTrack(CreateVideoTrack(track_label), stream_ids); |
| ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type()); |
| } |
| |
| void AddVideoStream(const std::string& label) { |
| rtc::scoped_refptr<MediaStreamInterface> stream( |
| pc_factory_->CreateLocalMediaStream(label)); |
| stream->AddTrack(CreateVideoTrack(label + "v0")); |
| ASSERT_TRUE(pc_->AddStream(stream)); |
| } |
| |
| rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack( |
| const std::string& label) { |
| return pc_factory_->CreateAudioTrack(label, nullptr); |
| } |
| |
| void AddAudioTrack(const std::string& track_label, |
| const std::vector<std::string>& stream_ids = {}) { |
| auto sender_or_error = |
| pc_->AddTrack(CreateAudioTrack(track_label), stream_ids); |
| ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type()); |
| } |
| |
| void AddAudioStream(const std::string& label) { |
| rtc::scoped_refptr<MediaStreamInterface> stream( |
| pc_factory_->CreateLocalMediaStream(label)); |
| stream->AddTrack(CreateAudioTrack(label + "a0")); |
| ASSERT_TRUE(pc_->AddStream(stream)); |
| } |
| |
| void AddAudioVideoStream(const std::string& stream_id, |
| const std::string& audio_track_label, |
| const std::string& video_track_label) { |
| // Create a local stream. |
| rtc::scoped_refptr<MediaStreamInterface> stream( |
| pc_factory_->CreateLocalMediaStream(stream_id)); |
| stream->AddTrack(CreateAudioTrack(audio_track_label)); |
| stream->AddTrack(CreateVideoTrack(video_track_label)); |
| ASSERT_TRUE(pc_->AddStream(stream)); |
| } |
| |
| rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType( |
| cricket::MediaType media_type) { |
| for (auto receiver : pc_->GetReceivers()) { |
| if (receiver->media_type() == media_type) { |
| return receiver; |
| } |
| } |
| return nullptr; |
| } |
| |
| bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc, |
| const RTCOfferAnswerOptions* options, |
| bool offer) { |
| rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); |
| if (offer) { |
| pc_->CreateOffer(observer, options ? *options : RTCOfferAnswerOptions()); |
| } else { |
| pc_->CreateAnswer(observer, options ? *options : RTCOfferAnswerOptions()); |
| } |
| EXPECT_EQ_WAIT(true, observer->called(), kTimeout); |
| *desc = observer->MoveDescription(); |
| return observer->result(); |
| } |
| |
| bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc, |
| const RTCOfferAnswerOptions* options) { |
| return DoCreateOfferAnswer(desc, options, true); |
| } |
| |
| bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc, |
| const RTCOfferAnswerOptions* options) { |
| return DoCreateOfferAnswer(desc, options, false); |
| } |
| |
| bool DoSetSessionDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc, |
| bool local) { |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); |
| if (local) { |
| pc_->SetLocalDescription(observer, desc.release()); |
| } else { |
| pc_->SetRemoteDescription(observer, desc.release()); |
| } |
| if (pc_->signaling_state() != PeerConnectionInterface::kClosed) { |
| EXPECT_EQ_WAIT(true, observer->called(), kTimeout); |
| } |
| return observer->result(); |
| } |
| |
| bool DoSetLocalDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc) { |
| return DoSetSessionDescription(std::move(desc), true); |
| } |
| |
| bool DoSetRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc) { |
| return DoSetSessionDescription(std::move(desc), false); |
| } |
| |
| // Calls PeerConnection::GetStats and check the return value. |
| // It does not verify the values in the StatReports since a RTCP packet might |
| // be required. |
| bool DoGetStats(MediaStreamTrackInterface* track) { |
| rtc::scoped_refptr<MockStatsObserver> observer( |
| new rtc::RefCountedObject<MockStatsObserver>()); |
| if (!pc_->GetStats(observer, track, |
| PeerConnectionInterface::kStatsOutputLevelStandard)) |
| return false; |
| EXPECT_TRUE_WAIT(observer->called(), kTimeout); |
| return observer->called(); |
| } |
| |
| // Call the standards-compliant GetStats function. |
| bool DoGetRTCStats() { |
| rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback( |
| new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>()); |
| pc_->GetStats(callback); |
| EXPECT_TRUE_WAIT(callback->called(), kTimeout); |
| return callback->called(); |
| } |
| |
| void InitiateCall() { |
| CreatePeerConnectionWithoutDtls(); |
| // Create a local stream with audio&video tracks. |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| AddAudioVideoStream(kStreamId1, "audio_track", "video_track"); |
| } else { |
| // Unified Plan does not support AddStream, so just add an audio and video |
| // track. |
| AddAudioTrack(kAudioTracks[0], {kStreamId1}); |
| AddVideoTrack(kVideoTracks[0], {kStreamId1}); |
| } |
| CreateOfferReceiveAnswer(); |
| } |
| |
| // Verify that RTP Header extensions has been negotiated for audio and video. |
| void VerifyRemoteRtpHeaderExtensions() { |
| const cricket::MediaContentDescription* desc = |
| cricket::GetFirstAudioContentDescription( |
| pc_->remote_description()->description()); |
| ASSERT_TRUE(desc != NULL); |
| EXPECT_GT(desc->rtp_header_extensions().size(), 0u); |
| |
| desc = cricket::GetFirstVideoContentDescription( |
| pc_->remote_description()->description()); |
| ASSERT_TRUE(desc != NULL); |
| EXPECT_GT(desc->rtp_header_extensions().size(), 0u); |
| } |
| |
| void CreateOfferAsRemoteDescription() { |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| std::string sdp; |
| EXPECT_TRUE(offer->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> remote_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); |
| } |
| |
| void CreateAndSetRemoteOffer(const std::string& sdp) { |
| std::unique_ptr<SessionDescriptionInterface> remote_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); |
| } |
| |
| void CreateAnswerAsLocalDescription() { |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| ASSERT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| |
| // TODO(perkj): Currently SetLocalDescription fails if any parameters in an |
| // audio codec change, even if the parameter has nothing to do with |
| // receiving. Not all parameters are serialized to SDP. |
| // Since CreatePrAnswerAsLocalDescription serialize/deserialize |
| // the SessionDescription, it is necessary to do that here to in order to |
| // get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass. |
| // https://code.google.com/p/webrtc/issues/detail?id=1356 |
| std::string sdp; |
| EXPECT_TRUE(answer->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> new_answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); |
| EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer))); |
| EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); |
| } |
| |
| void CreatePrAnswerAsLocalDescription() { |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| ASSERT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| |
| std::string sdp; |
| EXPECT_TRUE(answer->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> pr_answer( |
| webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); |
| EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_); |
| } |
| |
| void CreateOfferReceiveAnswer() { |
| CreateOfferAsLocalDescription(); |
| std::string sdp; |
| EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); |
| CreateAnswerAsRemoteDescription(sdp); |
| } |
| |
| void CreateOfferAsLocalDescription() { |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| // TODO(perkj): Currently SetLocalDescription fails if any parameters in an |
| // audio codec change, even if the parameter has nothing to do with |
| // receiving. Not all parameters are serialized to SDP. |
| // Since CreatePrAnswerAsLocalDescription serialize/deserialize |
| // the SessionDescription, it is necessary to do that here to in order to |
| // get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass. |
| // https://code.google.com/p/webrtc/issues/detail?id=1356 |
| std::string sdp; |
| EXPECT_TRUE(offer->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> new_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| |
| EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_); |
| // Wait for the ice_complete message, so that SDP will have candidates. |
| EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout); |
| } |
| |
| void CreateAnswerAsRemoteDescription(const std::string& sdp) { |
| std::unique_ptr<SessionDescriptionInterface> answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); |
| ASSERT_TRUE(answer); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(answer))); |
| EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); |
| } |
| |
| void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) { |
| std::unique_ptr<SessionDescriptionInterface> pr_answer( |
| webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); |
| ASSERT_TRUE(pr_answer); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_); |
| std::unique_ptr<SessionDescriptionInterface> answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); |
| ASSERT_TRUE(answer); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(answer))); |
| EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); |
| } |
| |
| // Waits until a remote stream with the given id is signaled. This helper |
| // function will verify both OnAddTrack and OnAddStream (Plan B only) are |
| // called with the given stream id and expected number of tracks. |
| void WaitAndVerifyOnAddStream(const std::string& stream_id, |
| int expected_num_tracks) { |
| // Verify that both OnAddStream and OnAddTrack are called. |
| EXPECT_EQ_WAIT(stream_id, observer_.GetLastAddedStreamId(), kTimeout); |
| EXPECT_EQ_WAIT(expected_num_tracks, |
| observer_.CountAddTrackEventsForStream(stream_id), kTimeout); |
| } |
| |
| // Creates an offer and applies it as a local session description. |
| // Creates an answer with the same SDP an the offer but removes all lines |
| // that start with a:ssrc" |
| void CreateOfferReceiveAnswerWithoutSsrc() { |
| CreateOfferAsLocalDescription(); |
| std::string sdp; |
| EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); |
| SetSsrcToZero(&sdp); |
| CreateAnswerAsRemoteDescription(sdp); |
| } |
| |
| // This function creates a MediaStream with label kStreams[0] and |
| // |number_of_audio_tracks| and |number_of_video_tracks| tracks and the |
| // corresponding SessionDescriptionInterface. The SessionDescriptionInterface |
| // is returned and the MediaStream is stored in |
| // |reference_collection_| |
| std::unique_ptr<SessionDescriptionInterface> |
| CreateSessionDescriptionAndReference(size_t number_of_audio_tracks, |
| size_t number_of_video_tracks) { |
| EXPECT_LE(number_of_audio_tracks, 2u); |
| EXPECT_LE(number_of_video_tracks, 2u); |
| |
| reference_collection_ = StreamCollection::Create(); |
| std::string sdp_ms1 = std::string(kSdpStringInit); |
| |
| std::string mediastream_id = kStreams[0]; |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream( |
| webrtc::MediaStream::Create(mediastream_id)); |
| reference_collection_->AddStream(stream); |
| |
| if (number_of_audio_tracks > 0) { |
| sdp_ms1 += std::string(kSdpStringAudio); |
| sdp_ms1 += std::string(kSdpStringMs1Audio0); |
| AddAudioTrack(kAudioTracks[0], stream); |
| } |
| if (number_of_audio_tracks > 1) { |
| sdp_ms1 += kSdpStringMs1Audio1; |
| AddAudioTrack(kAudioTracks[1], stream); |
| } |
| |
| if (number_of_video_tracks > 0) { |
| sdp_ms1 += std::string(kSdpStringVideo); |
| sdp_ms1 += std::string(kSdpStringMs1Video0); |
| AddVideoTrack(kVideoTracks[0], stream); |
| } |
| if (number_of_video_tracks > 1) { |
| sdp_ms1 += kSdpStringMs1Video1; |
| AddVideoTrack(kVideoTracks[1], stream); |
| } |
| |
| return std::unique_ptr<SessionDescriptionInterface>( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1)); |
| } |
| |
| void AddAudioTrack(const std::string& track_id, |
| MediaStreamInterface* stream) { |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( |
| webrtc::AudioTrack::Create(track_id, nullptr)); |
| ASSERT_TRUE(stream->AddTrack(audio_track)); |
| } |
| |
| void AddVideoTrack(const std::string& track_id, |
| MediaStreamInterface* stream) { |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( |
| webrtc::VideoTrack::Create(track_id, |
| webrtc::FakeVideoTrackSource::Create(), |
| rtc::Thread::Current())); |
| ASSERT_TRUE(stream->AddTrack(video_track)); |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioTrack() { |
| CreatePeerConnectionWithoutDtls(); |
| AddAudioTrack(kAudioTracks[0]); |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| EXPECT_TRUE(DoCreateOffer(&offer, nullptr)); |
| return offer; |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() { |
| CreatePeerConnectionWithoutDtls(); |
| AddAudioStream(kStreamId1); |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| EXPECT_TRUE(DoCreateOffer(&offer, nullptr)); |
| return offer; |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> CreateAnswerWithOneAudioTrack() { |
| EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioTrack())); |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| EXPECT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| return answer; |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> |
| CreateAnswerWithOneAudioStream() { |
| EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioStream())); |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| EXPECT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| return answer; |
| } |
| |
| const std::string& GetFirstAudioStreamCname( |
| const SessionDescriptionInterface* desc) { |
| const cricket::AudioContentDescription* audio_desc = |
| cricket::GetFirstAudioContentDescription(desc->description()); |
| return audio_desc->streams()[0].cname; |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOptions( |
| const RTCOfferAnswerOptions& offer_answer_options) { |
| RTC_DCHECK(pc_); |
| rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); |
| pc_->CreateOffer(observer, offer_answer_options); |
| EXPECT_EQ_WAIT(true, observer->called(), kTimeout); |
| return observer->MoveDescription(); |
| } |
| |
| void CreateOfferWithOptionsAsRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface>* desc, |
| const RTCOfferAnswerOptions& offer_answer_options) { |
| *desc = CreateOfferWithOptions(offer_answer_options); |
| ASSERT_TRUE(desc != nullptr); |
| std::string sdp; |
| EXPECT_TRUE((*desc)->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> remote_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); |
| } |
| |
| void CreateOfferWithOptionsAsLocalDescription( |
| std::unique_ptr<SessionDescriptionInterface>* desc, |
| const RTCOfferAnswerOptions& offer_answer_options) { |
| *desc = CreateOfferWithOptions(offer_answer_options); |
| ASSERT_TRUE(desc != nullptr); |
| std::string sdp; |
| EXPECT_TRUE((*desc)->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> new_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| |
| EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer))); |
| EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_); |
| } |
| |
| bool HasCNCodecs(const cricket::ContentInfo* content) { |
| RTC_DCHECK(content); |
| RTC_DCHECK(content->media_description()); |
| for (const cricket::AudioCodec& codec : |
| content->media_description()->as_audio()->codecs()) { |
| if (codec.name == "CN") { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| const char* GetSdpStringWithStream1() const { |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| return kSdpStringWithStream1PlanB; |
| } else { |
| return kSdpStringWithStream1UnifiedPlan; |
| } |
| } |
| |
| const char* GetSdpStringWithStream1And2() const { |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| return kSdpStringWithStream1And2PlanB; |
| } else { |
| return kSdpStringWithStream1And2UnifiedPlan; |
| } |
| } |
| |
| std::unique_ptr<rtc::VirtualSocketServer> vss_; |
| rtc::AutoSocketServerThread main_; |
| rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| cricket::FakePortAllocator* port_allocator_ = nullptr; |
| FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr; |
| rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_; |
| rtc::scoped_refptr<PeerConnectionFactoryForTest> pc_factory_for_test_; |
| rtc::scoped_refptr<PeerConnectionInterface> pc_; |
| MockPeerConnectionObserver observer_; |
| rtc::scoped_refptr<StreamCollection> reference_collection_; |
| const SdpSemantics sdp_semantics_; |
| }; |
| |
| class PeerConnectionInterfaceTest |
| : public PeerConnectionInterfaceBaseTest, |
| public ::testing::WithParamInterface<SdpSemantics> { |
| protected: |
| PeerConnectionInterfaceTest() : PeerConnectionInterfaceBaseTest(GetParam()) {} |
| }; |
| |
| class PeerConnectionInterfaceTestPlanB |
| : public PeerConnectionInterfaceBaseTest { |
| protected: |
| PeerConnectionInterfaceTestPlanB() |
| : PeerConnectionInterfaceBaseTest(SdpSemantics::kPlanB) {} |
| }; |
| |
| // Generate different CNAMEs when PeerConnections are created. |
| // The CNAMEs are expected to be generated randomly. It is possible |
| // that the test fails, though the possibility is very low. |
| TEST_P(PeerConnectionInterfaceTest, CnameGenerationInOffer) { |
| std::unique_ptr<SessionDescriptionInterface> offer1 = |
| CreateOfferWithOneAudioTrack(); |
| std::unique_ptr<SessionDescriptionInterface> offer2 = |
| CreateOfferWithOneAudioTrack(); |
| EXPECT_NE(GetFirstAudioStreamCname(offer1.get()), |
| GetFirstAudioStreamCname(offer2.get())); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, CnameGenerationInAnswer) { |
| std::unique_ptr<SessionDescriptionInterface> answer1 = |
| CreateAnswerWithOneAudioTrack(); |
| std::unique_ptr<SessionDescriptionInterface> answer2 = |
| CreateAnswerWithOneAudioTrack(); |
| EXPECT_NE(GetFirstAudioStreamCname(answer1.get()), |
| GetFirstAudioStreamCname(answer2.get())); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, |
| CreatePeerConnectionWithDifferentConfigurations) { |
| CreatePeerConnectionWithDifferentConfigurations(); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, |
| CreatePeerConnectionWithDifferentIceTransportsTypes) { |
| CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone); |
| EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter()); |
| CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kRelay); |
| EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter()); |
| CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNoHost); |
| EXPECT_EQ(cricket::CF_ALL & ~cricket::CF_HOST, |
| port_allocator_->candidate_filter()); |
| CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kAll); |
| EXPECT_EQ(cricket::CF_ALL, port_allocator_->candidate_filter()); |
| } |
| |
| // Test that when a PeerConnection is created with a nonzero candidate pool |
| // size, the pooled PortAllocatorSession is created with all the attributes |
| // in the RTCConfiguration. |
| TEST_P(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) { |
| PeerConnectionInterface::RTCConfiguration config; |
| PeerConnectionInterface::IceServer server; |
| server.uri = kStunAddressOnly; |
| config.servers.push_back(server); |
| config.type = PeerConnectionInterface::kRelay; |
| config.disable_ipv6 = true; |
| config.tcp_candidate_policy = |
| PeerConnectionInterface::kTcpCandidatePolicyDisabled; |
| config.candidate_network_policy = |
| PeerConnectionInterface::kCandidateNetworkPolicyLowCost; |
| config.ice_candidate_pool_size = 1; |
| CreatePeerConnection(config); |
| |
| const cricket::FakePortAllocatorSession* session = |
| static_cast<const cricket::FakePortAllocatorSession*>( |
| port_allocator_->GetPooledSession()); |
| ASSERT_NE(nullptr, session); |
| EXPECT_EQ(1UL, session->stun_servers().size()); |
| EXPECT_EQ(0U, session->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6); |
| EXPECT_LT(0U, session->flags() & cricket::PORTALLOCATOR_DISABLE_TCP); |
| EXPECT_LT(0U, |
| session->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); |
| } |
| |
| // Test that network-related RTCConfiguration members are applied to the |
| // PortAllocator when CreatePeerConnection is called. Specifically: |
| // - disable_ipv6_on_wifi |
| // - max_ipv6_networks |
| // - tcp_candidate_policy |
| // - candidate_network_policy |
| // - prune_turn_ports |
| // |
| // Note that the candidate filter (RTCConfiguration::type) is already tested |
| // above. |
| TEST_P(PeerConnectionInterfaceTest, |
| CreatePeerConnectionAppliesNetworkConfigToPortAllocator) { |
| // Create fake port allocator. |
| std::unique_ptr<cricket::FakePortAllocator> port_allocator( |
| new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr)); |
| cricket::FakePortAllocator* raw_port_allocator = port_allocator.get(); |
| |
| // Create RTCConfiguration with some network-related fields relevant to |
| // PortAllocator populated. |
| PeerConnectionInterface::RTCConfiguration config; |
| config.disable_ipv6_on_wifi = true; |
| config.max_ipv6_networks = 10; |
| config.tcp_candidate_policy = |
| PeerConnectionInterface::kTcpCandidatePolicyDisabled; |
| config.candidate_network_policy = |
| PeerConnectionInterface::kCandidateNetworkPolicyLowCost; |
| config.prune_turn_ports = true; |
| |
| // Create the PC factory and PC with the above config. |
| rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory( |
| webrtc::CreatePeerConnectionFactory( |
| rtc::Thread::Current(), rtc::Thread::Current(), |
| rtc::Thread::Current(), fake_audio_capture_module_, |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| webrtc::CreateBuiltinVideoEncoderFactory(), |
| webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */, |
| nullptr /* audio_processing */)); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| pc_factory->CreatePeerConnection(config, std::move(port_allocator), |
| nullptr, &observer_)); |
| EXPECT_TRUE(pc.get()); |
| observer_.SetPeerConnectionInterface(pc.get()); |
| |
| // Now validate that the config fields set above were applied to the |
| // PortAllocator, as flags or otherwise. |
| EXPECT_FALSE(raw_port_allocator->flags() & |
| cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); |
| EXPECT_EQ(10, raw_port_allocator->max_ipv6_networks()); |
| EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP); |
| EXPECT_TRUE(raw_port_allocator->flags() & |
| cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); |
| EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY, |
| raw_port_allocator->turn_port_prune_policy()); |
| } |
| |
| // Check that GetConfiguration returns the configuration the PeerConnection was |
| // constructed with, before SetConfiguration is called. |
| TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.type = PeerConnectionInterface::kRelay; |
| CreatePeerConnection(config); |
| |
| PeerConnectionInterface::RTCConfiguration returned_config = |
| pc_->GetConfiguration(); |
| EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type); |
| } |
| |
| // Check that GetConfiguration returns the last configuration passed into |
| // SetConfiguration. |
| TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) { |
| PeerConnectionInterface::RTCConfiguration starting_config; |
| starting_config.bundle_policy = |
| webrtc::PeerConnection::kBundlePolicyMaxBundle; |
| CreatePeerConnection(starting_config); |
| |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| config.type = PeerConnectionInterface::kRelay; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| |
| PeerConnectionInterface::RTCConfiguration returned_config = |
| pc_->GetConfiguration(); |
| EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) { |
| CreatePeerConnection(); |
| |
| pc_->Close(); |
| |
| EXPECT_FALSE( |
| pc_->SetConfiguration(PeerConnectionInterface::RTCConfiguration()).ok()); |
| } |
| |
| TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) { |
| CreatePeerConnectionWithoutDtls(); |
| AddVideoStream(kStreamId1); |
| AddAudioStream(kStreamId2); |
| ASSERT_EQ(2u, pc_->local_streams()->count()); |
| |
| // Test we can add multiple local streams to one peerconnection. |
| rtc::scoped_refptr<MediaStreamInterface> stream( |
| pc_factory_->CreateLocalMediaStream(kStreamId3)); |
| rtc::scoped_refptr<AudioTrackInterface> audio_track( |
| pc_factory_->CreateAudioTrack(kStreamId3, |
| static_cast<AudioSourceInterface*>(NULL))); |
| stream->AddTrack(audio_track.get()); |
| EXPECT_TRUE(pc_->AddStream(stream)); |
| EXPECT_EQ(3u, pc_->local_streams()->count()); |
| |
| // Remove the third stream. |
| pc_->RemoveStream(pc_->local_streams()->at(2)); |
| EXPECT_EQ(2u, pc_->local_streams()->count()); |
| |
| // Remove the second stream. |
| pc_->RemoveStream(pc_->local_streams()->at(1)); |
| EXPECT_EQ(1u, pc_->local_streams()->count()); |
| |
| // Remove the first stream. |
| pc_->RemoveStream(pc_->local_streams()->at(0)); |
| EXPECT_EQ(0u, pc_->local_streams()->count()); |
| } |
| |
| // Test that the created offer includes streams we added. |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, AddedStreamsPresentInOffer) { |
| CreatePeerConnectionWithoutDtls(); |
| AddAudioVideoStream(kStreamId1, "audio_track", "video_track"); |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| const cricket::AudioContentDescription* audio_desc = |
| cricket::GetFirstAudioContentDescription(offer->description()); |
| EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track")); |
| |
| const cricket::VideoContentDescription* video_desc = |
| cricket::GetFirstVideoContentDescription(offer->description()); |
| EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track")); |
| |
| // Add another stream and ensure the offer includes both the old and new |
| // streams. |
| AddAudioVideoStream(kStreamId2, "audio_track2", "video_track2"); |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| audio_desc = cricket::GetFirstAudioContentDescription(offer->description()); |
| EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track")); |
| EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId2, "audio_track2")); |
| |
| video_desc = cricket::GetFirstVideoContentDescription(offer->description()); |
| EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track")); |
| EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId2, "video_track2")); |
| } |
| |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, RemoveStream) { |
| CreatePeerConnectionWithoutDtls(); |
| AddVideoStream(kStreamId1); |
| ASSERT_EQ(1u, pc_->local_streams()->count()); |
| pc_->RemoveStream(pc_->local_streams()->at(0)); |
| EXPECT_EQ(0u, pc_->local_streams()->count()); |
| } |
| |
| // Test for AddTrack and RemoveTrack methods. |
| // Tests that the created offer includes tracks we added, |
| // and that the RtpSenders are created correctly. |
| // Also tests that RemoveTrack removes the tracks from subsequent offers. |
| // Only tested with Plan B since Unified Plan is covered in more detail by tests |
| // in peerconnection_jsep_unittests.cc |
| TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackRemoveTrack) { |
| CreatePeerConnectionWithoutDtls(); |
| rtc::scoped_refptr<AudioTrackInterface> audio_track( |
| CreateAudioTrack("audio_track")); |
| rtc::scoped_refptr<VideoTrackInterface> video_track( |
| CreateVideoTrack("video_track")); |
| auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue(); |
| auto video_sender = pc_->AddTrack(video_track, {kStreamId1}).MoveValue(); |
| EXPECT_EQ(1UL, audio_sender->stream_ids().size()); |
| EXPECT_EQ(kStreamId1, audio_sender->stream_ids()[0]); |
| EXPECT_EQ("audio_track", audio_sender->id()); |
| EXPECT_EQ(audio_track, audio_sender->track()); |
| EXPECT_EQ(1UL, video_sender->stream_ids().size()); |
| EXPECT_EQ(kStreamId1, video_sender->stream_ids()[0]); |
| EXPECT_EQ("video_track", video_sender->id()); |
| EXPECT_EQ(video_track, video_sender->track()); |
| |
| // Now create an offer and check for the senders. |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| const cricket::ContentInfo* audio_content = |
| cricket::GetFirstAudioContent(offer->description()); |
| EXPECT_TRUE(ContainsTrack(audio_content->media_description()->streams(), |
| kStreamId1, "audio_track")); |
| |
| const cricket::ContentInfo* video_content = |
| cricket::GetFirstVideoContent(offer->description()); |
| EXPECT_TRUE(ContainsTrack(video_content->media_description()->streams(), |
| kStreamId1, "video_track")); |
| |
| EXPECT_TRUE(DoSetLocalDescription(std::move(offer))); |
| |
| // Now try removing the tracks. |
| EXPECT_TRUE(pc_->RemoveTrack(audio_sender)); |
| EXPECT_TRUE(pc_->RemoveTrack(video_sender)); |
| |
| // Create a new offer and ensure it doesn't contain the removed senders. |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| audio_content = cricket::GetFirstAudioContent(offer->description()); |
| EXPECT_FALSE(ContainsTrack(audio_content->media_description()->streams(), |
| kStreamId1, "audio_track")); |
| |
| video_content = cricket::GetFirstVideoContent(offer->description()); |
| EXPECT_FALSE(ContainsTrack(video_content->media_description()->streams(), |
| kStreamId1, "video_track")); |
| |
| EXPECT_TRUE(DoSetLocalDescription(std::move(offer))); |
| |
| // Calling RemoveTrack on a sender no longer attached to a PeerConnection |
| // should return false. |
| EXPECT_FALSE(pc_->RemoveTrack(audio_sender)); |
| EXPECT_FALSE(pc_->RemoveTrack(video_sender)); |
| } |
| |
| // Test creating senders without a stream specified, |
| // expecting a random stream ID to be generated. |
| TEST_P(PeerConnectionInterfaceTest, AddTrackWithoutStream) { |
| CreatePeerConnectionWithoutDtls(); |
| rtc::scoped_refptr<AudioTrackInterface> audio_track( |
| CreateAudioTrack("audio_track")); |
| rtc::scoped_refptr<VideoTrackInterface> video_track( |
| CreateVideoTrack("video_track")); |
| auto audio_sender = |
| pc_->AddTrack(audio_track, std::vector<std::string>()).MoveValue(); |
| auto video_sender = |
| pc_->AddTrack(video_track, std::vector<std::string>()).MoveValue(); |
| EXPECT_EQ("audio_track", audio_sender->id()); |
| EXPECT_EQ(audio_track, audio_sender->track()); |
| EXPECT_EQ("video_track", video_sender->id()); |
| EXPECT_EQ(video_track, video_sender->track()); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| // If the ID is truly a random GUID, it should be infinitely unlikely they |
| // will be the same. |
| EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids()); |
| } else { |
| // We allows creating tracks without stream ids under Unified Plan |
| // semantics. |
| EXPECT_EQ(0u, video_sender->stream_ids().size()); |
| EXPECT_EQ(0u, audio_sender->stream_ids().size()); |
| } |
| } |
| |
| // Test that we can call GetStats() after AddTrack but before connecting |
| // the PeerConnection to a peer. |
| TEST_P(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) { |
| CreatePeerConnectionWithoutDtls(); |
| rtc::scoped_refptr<AudioTrackInterface> audio_track( |
| CreateAudioTrack("audio_track")); |
| rtc::scoped_refptr<VideoTrackInterface> video_track( |
| CreateVideoTrack("video_track")); |
| auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>()); |
| auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>()); |
| EXPECT_TRUE(DoGetStats(nullptr)); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) { |
| CreatePeerConnectionWithoutDtls(); |
| rtc::scoped_refptr<AudioTrackInterface> audio_track( |
| CreateAudioTrack("audio_track")); |
| rtc::scoped_refptr<VideoTrackInterface> video_track( |
| CreateVideoTrack("video_track")); |
| auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>()); |
| ASSERT_TRUE(audio_sender.ok()); |
| auto* audio_sender_proxy = |
| static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>( |
| audio_sender.value().get()); |
| EXPECT_NE(0, audio_sender_proxy->internal()->AttachmentId()); |
| |
| auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>()); |
| ASSERT_TRUE(video_sender.ok()); |
| auto* video_sender_proxy = |
| static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>( |
| video_sender.value().get()); |
| EXPECT_NE(0, video_sender_proxy->internal()->AttachmentId()); |
| } |
| |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) { |
| CreatePeerConnectionWithoutDtls(); |
| AddVideoStream(kStreamId1); |
| auto senders = pc_->GetSenders(); |
| ASSERT_EQ(1u, senders.size()); |
| auto* sender_proxy = |
| static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>( |
| senders[0].get()); |
| EXPECT_NE(0, sender_proxy->internal()->AttachmentId()); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) { |
| InitiateCall(); |
| WaitAndVerifyOnAddStream(kStreamId1, 2); |
| VerifyRemoteRtpHeaderExtensions(); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) { |
| CreatePeerConnectionWithoutDtls(); |
| AddVideoTrack(kVideoTracks[0], {kStreamId1}); |
| CreateOfferAsLocalDescription(); |
| std::string offer; |
| EXPECT_TRUE(pc_->local_description()->ToString(&offer)); |
| CreatePrAnswerAndAnswerAsRemoteDescription(offer); |
| WaitAndVerifyOnAddStream(kStreamId1, 1); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) { |
| CreatePeerConnectionWithoutDtls(); |
| AddVideoTrack(kVideoTracks[0], {kStreamId1}); |
| |
| CreateOfferAsRemoteDescription(); |
| CreateAnswerAsLocalDescription(); |
| |
| WaitAndVerifyOnAddStream(kStreamId1, 1); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) { |
| CreatePeerConnectionWithoutDtls(); |
| AddVideoTrack(kVideoTracks[0], {kStreamId1}); |
| |
| CreateOfferAsRemoteDescription(); |
| CreatePrAnswerAsLocalDescription(); |
| CreateAnswerAsLocalDescription(); |
| |
| WaitAndVerifyOnAddStream(kStreamId1, 1); |
| } |
| |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, Renegotiate) { |
| InitiateCall(); |
| ASSERT_EQ(1u, pc_->remote_streams()->count()); |
| pc_->RemoveStream(pc_->local_streams()->at(0)); |
| CreateOfferReceiveAnswer(); |
| EXPECT_EQ(0u, pc_->remote_streams()->count()); |
| AddVideoStream(kStreamId1); |
| CreateOfferReceiveAnswer(); |
| } |
| |
| // Tests that after negotiating an audio only call, the respondent can perform a |
| // renegotiation that removes the audio stream. |
| TEST_F(PeerConnectionInterfaceTestPlanB, RenegotiateAudioOnly) { |
| CreatePeerConnectionWithoutDtls(); |
| AddAudioStream(kStreamId1); |
| CreateOfferAsRemoteDescription(); |
| CreateAnswerAsLocalDescription(); |
| |
| ASSERT_EQ(1u, pc_->remote_streams()->count()); |
| pc_->RemoveStream(pc_->local_streams()->at(0)); |
| CreateOfferReceiveAnswer(); |
| EXPECT_EQ(0u, pc_->remote_streams()->count()); |
| } |
| |
| // Test that candidates are generated and that we can parse our own candidates. |
| TEST_P(PeerConnectionInterfaceTest, IceCandidates) { |
| CreatePeerConnectionWithoutDtls(); |
| |
| EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate())); |
| // SetRemoteDescription takes ownership of offer. |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| AddVideoTrack(kVideoTracks[0]); |
| EXPECT_TRUE(DoCreateOffer(&offer, nullptr)); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(offer))); |
| |
| // SetLocalDescription takes ownership of answer. |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| EXPECT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| EXPECT_TRUE(DoSetLocalDescription(std::move(answer))); |
| |
| EXPECT_TRUE_WAIT(observer_.last_candidate() != nullptr, kTimeout); |
| EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout); |
| |
| EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate())); |
| } |
| |
| // Test that CreateOffer and CreateAnswer will fail if the track labels are |
| // not unique. |
| TEST_F(PeerConnectionInterfaceTestPlanB, CreateOfferAnswerWithInvalidStream) { |
| CreatePeerConnectionWithoutDtls(); |
| // Create a regular offer for the CreateAnswer test later. |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| EXPECT_TRUE(DoCreateOffer(&offer, nullptr)); |
| EXPECT_TRUE(offer); |
| offer.reset(); |
| |
| // Create a local stream with audio&video tracks having same label. |
| AddAudioTrack("track_label", {kStreamId1}); |
| AddVideoTrack("track_label", {kStreamId1}); |
| |
| // Test CreateOffer |
| EXPECT_FALSE(DoCreateOffer(&offer, nullptr)); |
| |
| // Test CreateAnswer |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| EXPECT_FALSE(DoCreateAnswer(&answer, nullptr)); |
| } |
| |
| // Test that we will get different SSRCs for each tracks in the offer and answer |
| // we created. |
| TEST_P(PeerConnectionInterfaceTest, SsrcInOfferAnswer) { |
| CreatePeerConnectionWithoutDtls(); |
| // Create a local stream with audio&video tracks having different labels. |
| AddAudioTrack(kAudioTracks[0], {kStreamId1}); |
| AddVideoTrack(kVideoTracks[0], {kStreamId1}); |
| |
| // Test CreateOffer |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| int audio_ssrc = 0; |
| int video_ssrc = 0; |
| EXPECT_TRUE( |
| GetFirstSsrc(GetFirstAudioContent(offer->description()), &audio_ssrc)); |
| EXPECT_TRUE( |
| GetFirstSsrc(GetFirstVideoContent(offer->description()), &video_ssrc)); |
| EXPECT_NE(audio_ssrc, video_ssrc); |
| |
| // Test CreateAnswer |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(offer))); |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| ASSERT_TRUE(DoCreateAnswer(&answer, nullptr)); |
| audio_ssrc = 0; |
| video_ssrc = 0; |
| EXPECT_TRUE( |
| GetFirstSsrc(GetFirstAudioContent(answer->description()), &audio_ssrc)); |
| EXPECT_TRUE( |
| GetFirstSsrc(GetFirstVideoContent(answer->description()), &video_ssrc)); |
| EXPECT_NE(audio_ssrc, video_ssrc); |
| } |
| |
| // Test that it's possible to call AddTrack on a MediaStream after adding |
| // the stream to a PeerConnection. |
| // TODO(deadbeef): Remove this test once this behavior is no longer supported. |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackAfterAddStream) { |
| CreatePeerConnectionWithoutDtls(); |
| // Create audio stream and add to PeerConnection. |
| AddAudioStream(kStreamId1); |
| MediaStreamInterface* stream = pc_->local_streams()->at(0); |
| |
| // Add video track to the audio-only stream. |
| rtc::scoped_refptr<VideoTrackInterface> video_track( |
| CreateVideoTrack("video_label")); |
| stream->AddTrack(video_track.get()); |
| |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| const cricket::MediaContentDescription* video_desc = |
| cricket::GetFirstVideoContentDescription(offer->description()); |
| EXPECT_TRUE(video_desc != nullptr); |
| } |
| |
| // Test that it's possible to call RemoveTrack on a MediaStream after adding |
| // the stream to a PeerConnection. |
| // TODO(deadbeef): Remove this test once this behavior is no longer supported. |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackAfterAddStream) { |
| CreatePeerConnectionWithoutDtls(); |
| // Create audio/video stream and add to PeerConnection. |
| AddAudioVideoStream(kStreamId1, "audio_label", "video_label"); |
| MediaStreamInterface* stream = pc_->local_streams()->at(0); |
| |
| // Remove the video track. |
| stream->RemoveTrack(stream->GetVideoTracks()[0]); |
| |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| const cricket::MediaContentDescription* video_desc = |
| cricket::GetFirstVideoContentDescription(offer->description()); |
| EXPECT_TRUE(video_desc == nullptr); |
| } |
| |
| // Test creating a sender with a stream ID, and ensure the ID is populated |
| // in the offer. |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, CreateSenderWithStream) { |
| CreatePeerConnectionWithoutDtls(); |
| pc_->CreateSender("video", kStreamId1); |
| |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); |
| |
| const cricket::MediaContentDescription* video_desc = |
| cricket::GetFirstVideoContentDescription(offer->description()); |
| ASSERT_TRUE(video_desc != nullptr); |
| ASSERT_EQ(1u, video_desc->streams().size()); |
| EXPECT_EQ(kStreamId1, video_desc->streams()[0].first_stream_id()); |
| } |
| |
| // Test that we can specify a certain track that we want statistics about. |
| TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) { |
| InitiateCall(); |
| ASSERT_LT(0u, pc_->GetSenders().size()); |
| ASSERT_LT(0u, pc_->GetReceivers().size()); |
| rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio = |
| pc_->GetReceivers()[0]->track(); |
| EXPECT_TRUE(DoGetStats(remote_audio)); |
| |
| // Remove the stream. Since we are sending to our selves the local |
| // and the remote stream is the same. |
| pc_->RemoveTrack(pc_->GetSenders()[0]); |
| // Do a re-negotiation. |
| CreateOfferReceiveAnswer(); |
| |
| // Test that we still can get statistics for the old track. Even if it is not |
| // sent any longer. |
| EXPECT_TRUE(DoGetStats(remote_audio)); |
| } |
| |
| // Test that we can get stats on a video track. |
| TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) { |
| InitiateCall(); |
| auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO); |
| ASSERT_TRUE(video_receiver); |
| EXPECT_TRUE(DoGetStats(video_receiver->track())); |
| } |
| |
| // Test that we don't get statistics for an invalid track. |
| TEST_P(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) { |
| InitiateCall(); |
| rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track( |
| pc_factory_->CreateAudioTrack("unknown track", NULL)); |
| EXPECT_FALSE(DoGetStats(unknown_audio_track)); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) { |
| CreatePeerConnectionWithoutDtls(); |
| EXPECT_TRUE(DoGetRTCStats()); |
| // Clearing stats cache is needed now, but should be temporary. |
| // https://bugs.chromium.org/p/webrtc/issues/detail?id=8693 |
| pc_->ClearStatsCache(); |
| AddAudioTrack(kAudioTracks[0], {kStreamId1}); |
| AddVideoTrack(kVideoTracks[0], {kStreamId1}); |
| EXPECT_TRUE(DoGetRTCStats()); |
| pc_->ClearStatsCache(); |
| CreateOfferReceiveAnswer(); |
| EXPECT_TRUE(DoGetRTCStats()); |
| } |
| |
| // This test setup two RTP data channels in loop back. |
| TEST_P(PeerConnectionInterfaceTest, TestDataChannel) { |
| RTCConfiguration config; |
| config.enable_rtp_data_channel = true; |
| config.enable_dtls_srtp = false; |
| CreatePeerConnection(config); |
| rtc::scoped_refptr<DataChannelInterface> data1 = |
| pc_->CreateDataChannel("test1", NULL); |
| rtc::scoped_refptr<DataChannelInterface> data2 = |
| pc_->CreateDataChannel("test2", NULL); |
| ASSERT_TRUE(data1 != NULL); |
| std::unique_ptr<MockDataChannelObserver> observer1( |
| new MockDataChannelObserver(data1)); |
| std::unique_ptr<MockDataChannelObserver> observer2( |
| new MockDataChannelObserver(data2)); |
| |
| EXPECT_EQ(DataChannelInterface::kConnecting, data1->state()); |
| EXPECT_EQ(DataChannelInterface::kConnecting, data2->state()); |
| std::string data_to_send1 = "testing testing"; |
| std::string data_to_send2 = "testing something else"; |
| EXPECT_FALSE(data1->Send(DataBuffer(data_to_send1))); |
| |
| CreateOfferReceiveAnswer(); |
| EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout); |
| EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout); |
| |
| EXPECT_EQ(DataChannelInterface::kOpen, data1->state()); |
| EXPECT_EQ(DataChannelInterface::kOpen, data2->state()); |
| EXPECT_TRUE(data1->Send(DataBuffer(data_to_send1))); |
| EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2))); |
| |
| EXPECT_EQ_WAIT(data_to_send1, observer1->last_message(), kTimeout); |
| EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout); |
| |
| data1->Close(); |
| EXPECT_EQ(DataChannelInterface::kClosing, data1->state()); |
| CreateOfferReceiveAnswer(); |
| EXPECT_FALSE(observer1->IsOpen()); |
| EXPECT_EQ(DataChannelInterface::kClosed, data1->state()); |
| EXPECT_TRUE(observer2->IsOpen()); |
| |
| data_to_send2 = "testing something else again"; |
| EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2))); |
| |
| EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout); |
| } |
| |
| // This test verifies that sendnig binary data over RTP data channels should |
| // fail. |
| TEST_P(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) { |
| RTCConfiguration config; |
| config.enable_rtp_data_channel = true; |
| config.enable_dtls_srtp = false; |
| CreatePeerConnection(config); |
| rtc::scoped_refptr<DataChannelInterface> data1 = |
| pc_->CreateDataChannel("test1", NULL); |
| rtc::scoped_refptr<DataChannelInterface> data2 = |
| pc_->CreateDataChannel("test2", NULL); |
| ASSERT_TRUE(data1 != NULL); |
| std::unique_ptr<MockDataChannelObserver> observer1( |
| new MockDataChannelObserver(data1)); |
| std::unique_ptr<MockDataChannelObserver> observer2( |
| new MockDataChannelObserver(data2)); |
| |
| EXPECT_EQ(DataChannelInterface::kConnecting, data1->state()); |
| EXPECT_EQ(DataChannelInterface::kConnecting, data2->state()); |
| |
| CreateOfferReceiveAnswer(); |
| EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout); |
| EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout); |
| |
| EXPECT_EQ(DataChannelInterface::kOpen, data1->state()); |
| EXPECT_EQ(DataChannelInterface::kOpen, data2->state()); |
| |
| rtc::CopyOnWriteBuffer buffer("test", 4); |
| EXPECT_FALSE(data1->Send(DataBuffer(buffer, true))); |
| } |
| |
| // This test setup a RTP data channels in loop back and test that a channel is |
| // opened even if the remote end answer with a zero SSRC. |
| TEST_P(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) { |
| RTCConfiguration config; |
| config.enable_rtp_data_channel = true; |
| config.enable_dtls_srtp = false; |
| CreatePeerConnection(config); |
| rtc::scoped_refptr<DataChannelInterface> data1 = |
| pc_->CreateDataChannel("test1", NULL); |
| std::unique_ptr<MockDataChannelObserver> observer1( |
| new MockDataChannelObserver(data1)); |
| |
| CreateOfferReceiveAnswerWithoutSsrc(); |
| |
| EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout); |
| |
| data1->Close(); |
| EXPECT_EQ(DataChannelInterface::kClosing, data1->state()); |
| CreateOfferReceiveAnswerWithoutSsrc(); |
| EXPECT_EQ(DataChannelInterface::kClosed, data1->state()); |
| EXPECT_FALSE(observer1->IsOpen()); |
| } |
| |
| // This test that if a data channel is added in an answer a receive only channel |
| // channel is created. |
| TEST_P(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) { |
| RTCConfiguration config; |
| config.enable_rtp_data_channel = true; |
| config.enable_dtls_srtp = false; |
| |
| CreatePeerConnection(config); |
| |
| std::string offer_label = "offer_channel"; |
| rtc::scoped_refptr<DataChannelInterface> offer_channel = |
| pc_->CreateDataChannel(offer_label, NULL); |
| |
| CreateOfferAsLocalDescription(); |
| |
| // Replace the data channel label in the offer and apply it as an answer. |
| std::string receive_label = "answer_channel"; |
| std::string sdp; |
| EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); |
| absl::StrReplaceAll({{offer_label, receive_label}}, &sdp); |
| CreateAnswerAsRemoteDescription(sdp); |
| |
| // Verify that a new incoming data channel has been created and that |
| // it is open but can't we written to. |
| ASSERT_TRUE(observer_.last_datachannel_ != NULL); |
| DataChannelInterface* received_channel = observer_.last_datachannel_; |
| EXPECT_EQ(DataChannelInterface::kConnecting, received_channel->state()); |
| EXPECT_EQ(receive_label, received_channel->label()); |
| EXPECT_FALSE(received_channel->Send(DataBuffer("something"))); |
| |
| // Verify that the channel we initially offered has been rejected. |
| EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state()); |
| |
| // Do another offer / answer exchange and verify that the data channel is |
| // opened. |
| CreateOfferReceiveAnswer(); |
| EXPECT_EQ_WAIT(DataChannelInterface::kOpen, received_channel->state(), |
| kTimeout); |
| } |
| |
| // This test that no data channel is returned if a reliable channel is |
| // requested. |
| // TODO(perkj): Remove this test once reliable channels are implemented. |
| TEST_P(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_rtp_data_channel = true; |
| CreatePeerConnection(rtc_config); |
| |
| std::string label = "test"; |
| webrtc::DataChannelInit config; |
| config.reliable = true; |
| rtc::scoped_refptr<DataChannelInterface> channel = |
| pc_->CreateDataChannel(label, &config); |
| EXPECT_TRUE(channel == NULL); |
| } |
| |
| // Verifies that duplicated label is not allowed for RTP data channel. |
| TEST_P(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) { |
| RTCConfiguration config; |
| config.enable_rtp_data_channel = true; |
| CreatePeerConnection(config); |
| |
| std::string label = "test"; |
| rtc::scoped_refptr<DataChannelInterface> channel = |
| pc_->CreateDataChannel(label, nullptr); |
| EXPECT_NE(channel, nullptr); |
| |
| rtc::scoped_refptr<DataChannelInterface> dup_channel = |
| pc_->CreateDataChannel(label, nullptr); |
| EXPECT_EQ(dup_channel, nullptr); |
| } |
| |
| // This tests that a SCTP data channel is returned using different |
| // DataChannelInit configurations. |
| TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| |
| webrtc::DataChannelInit config; |
| rtc::scoped_refptr<DataChannelInterface> channel = |
| pc_->CreateDataChannel("1", &config); |
| EXPECT_TRUE(channel != NULL); |
| EXPECT_TRUE(channel->reliable()); |
| EXPECT_TRUE(observer_.renegotiation_needed_); |
| observer_.renegotiation_needed_ = false; |
| |
| config.ordered = false; |
| channel = pc_->CreateDataChannel("2", &config); |
| EXPECT_TRUE(channel != NULL); |
| EXPECT_TRUE(channel->reliable()); |
| EXPECT_FALSE(observer_.renegotiation_needed_); |
| |
| config.ordered = true; |
| config.maxRetransmits = 0; |
| channel = pc_->CreateDataChannel("3", &config); |
| EXPECT_TRUE(channel != NULL); |
| EXPECT_FALSE(channel->reliable()); |
| EXPECT_FALSE(observer_.renegotiation_needed_); |
| |
| config.maxRetransmits = absl::nullopt; |
| config.maxRetransmitTime = 0; |
| channel = pc_->CreateDataChannel("4", &config); |
| EXPECT_TRUE(channel != NULL); |
| EXPECT_FALSE(channel->reliable()); |
| EXPECT_FALSE(observer_.renegotiation_needed_); |
| } |
| |
| // For backwards compatibility, we want people who "unset" maxRetransmits |
| // and maxRetransmitTime by setting them to -1 to get what they want. |
| TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| |
| webrtc::DataChannelInit config; |
| config.maxRetransmitTime = -1; |
| config.maxRetransmits = -1; |
| rtc::scoped_refptr<DataChannelInterface> channel = |
| pc_->CreateDataChannel("1", &config); |
| EXPECT_TRUE(channel != NULL); |
| } |
| |
| // This tests that no data channel is returned if both maxRetransmits and |
| // maxRetransmitTime are set for SCTP data channels. |
| TEST_P(PeerConnectionInterfaceTest, |
| CreateSctpDataChannelShouldFailForInvalidConfig) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| |
| std::string label = "test"; |
| webrtc::DataChannelInit config; |
| config.maxRetransmits = 0; |
| config.maxRetransmitTime = 0; |
| |
| rtc::scoped_refptr<DataChannelInterface> channel = |
| pc_->CreateDataChannel(label, &config); |
| EXPECT_TRUE(channel == NULL); |
| } |
| |
| // The test verifies that creating a SCTP data channel with an id already in use |
| // or out of range should fail. |
| TEST_P(PeerConnectionInterfaceTest, |
| CreateSctpDataChannelWithInvalidIdShouldFail) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| |
| webrtc::DataChannelInit config; |
| rtc::scoped_refptr<DataChannelInterface> channel; |
| |
| config.id = 1; |
| config.negotiated = true; |
| channel = pc_->CreateDataChannel("1", &config); |
| EXPECT_TRUE(channel != NULL); |
| EXPECT_EQ(1, channel->id()); |
| |
| channel = pc_->CreateDataChannel("x", &config); |
| EXPECT_TRUE(channel == NULL); |
| |
| config.id = cricket::kMaxSctpSid; |
| config.negotiated = true; |
| channel = pc_->CreateDataChannel("max", &config); |
| EXPECT_TRUE(channel != NULL); |
| EXPECT_EQ(config.id, channel->id()); |
| |
| config.id = cricket::kMaxSctpSid + 1; |
| config.negotiated = true; |
| channel = pc_->CreateDataChannel("x", &config); |
| EXPECT_TRUE(channel == NULL); |
| } |
| |
| // Verifies that duplicated label is allowed for SCTP data channel. |
| TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| |
| std::string label = "test"; |
| rtc::scoped_refptr<DataChannelInterface> channel = |
| pc_->CreateDataChannel(label, nullptr); |
| EXPECT_NE(channel, nullptr); |
| |
| rtc::scoped_refptr<DataChannelInterface> dup_channel = |
| pc_->CreateDataChannel(label, nullptr); |
| EXPECT_NE(dup_channel, nullptr); |
| } |
| |
| // This test verifies that OnRenegotiationNeeded is fired for every new RTP |
| // DataChannel. |
| TEST_P(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_rtp_data_channel = true; |
| rtc_config.enable_dtls_srtp = false; |
| CreatePeerConnection(rtc_config); |
| |
| rtc::scoped_refptr<DataChannelInterface> dc1 = |
| pc_->CreateDataChannel("test1", NULL); |
| EXPECT_TRUE(observer_.renegotiation_needed_); |
| observer_.renegotiation_needed_ = false; |
| |
| CreateOfferReceiveAnswer(); |
| |
| rtc::scoped_refptr<DataChannelInterface> dc2 = |
| pc_->CreateDataChannel("test2", NULL); |
| EXPECT_EQ(observer_.renegotiation_needed_, |
| GetParam() == SdpSemantics::kPlanB); |
| } |
| |
| // This test that a data channel closes when a PeerConnection is deleted/closed. |
| TEST_P(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_rtp_data_channel = true; |
| rtc_config.enable_dtls_srtp = false; |
| CreatePeerConnection(rtc_config); |
| |
| rtc::scoped_refptr<DataChannelInterface> data1 = |
| pc_->CreateDataChannel("test1", NULL); |
| rtc::scoped_refptr<DataChannelInterface> data2 = |
| pc_->CreateDataChannel("test2", NULL); |
| ASSERT_TRUE(data1 != NULL); |
| std::unique_ptr<MockDataChannelObserver> observer1( |
| new MockDataChannelObserver(data1)); |
| std::unique_ptr<MockDataChannelObserver> observer2( |
| new MockDataChannelObserver(data2)); |
| |
| CreateOfferReceiveAnswer(); |
| EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout); |
| EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout); |
| |
| ReleasePeerConnection(); |
| EXPECT_EQ(DataChannelInterface::kClosed, data1->state()); |
| EXPECT_EQ(DataChannelInterface::kClosed, data2->state()); |
| } |
| |
| // This tests that RTP data channels can be rejected in an answer. |
| TEST_P(PeerConnectionInterfaceTest, TestRejectRtpDataChannelInAnswer) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_rtp_data_channel = true; |
| rtc_config.enable_dtls_srtp = false; |
| CreatePeerConnection(rtc_config); |
| |
| rtc::scoped_refptr<DataChannelInterface> offer_channel( |
| pc_->CreateDataChannel("offer_channel", NULL)); |
| |
| CreateOfferAsLocalDescription(); |
| |
| // Create an answer where the m-line for data channels are rejected. |
| std::string sdp; |
| EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); |
| ASSERT_TRUE(answer); |
| cricket::ContentInfo* data_info = |
| cricket::GetFirstDataContent(answer->description()); |
| data_info->rejected = true; |
| |
| DoSetRemoteDescription(std::move(answer)); |
| EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state()); |
| } |
| |
| #ifdef WEBRTC_HAVE_SCTP |
| // This tests that SCTP data channels can be rejected in an answer. |
| TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer) |
| #else |
| TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer) |
| #endif |
| { |
| RTCConfiguration rtc_config; |
| CreatePeerConnection(rtc_config); |
| |
| rtc::scoped_refptr<DataChannelInterface> offer_channel( |
| pc_->CreateDataChannel("offer_channel", NULL)); |
| |
| CreateOfferAsLocalDescription(); |
| |
| // Create an answer where the m-line for data channels are rejected. |
| std::string sdp; |
| EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); |
| ASSERT_TRUE(answer); |
| cricket::ContentInfo* data_info = |
| cricket::GetFirstDataContent(answer->description()); |
| data_info->rejected = true; |
| |
| DoSetRemoteDescription(std::move(answer)); |
| EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state()); |
| } |
| |
| // Test that we can create a session description from an SDP string from |
| // FireFox, use it as a remote session description, generate an answer and use |
| // the answer as a local description. |
| TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| AddAudioTrack("audio_label"); |
| AddVideoTrack("video_label"); |
| std::unique_ptr<SessionDescriptionInterface> desc( |
| webrtc::CreateSessionDescription(SdpType::kOffer, |
| webrtc::kFireFoxSdpOffer, nullptr)); |
| EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false)); |
| CreateAnswerAsLocalDescription(); |
| ASSERT_TRUE(pc_->local_description() != NULL); |
| ASSERT_TRUE(pc_->remote_description() != NULL); |
| |
| const cricket::ContentInfo* content = |
| cricket::GetFirstAudioContent(pc_->local_description()->description()); |
| ASSERT_TRUE(content != NULL); |
| EXPECT_FALSE(content->rejected); |
| |
| content = |
| cricket::GetFirstVideoContent(pc_->local_description()->description()); |
| ASSERT_TRUE(content != NULL); |
| EXPECT_FALSE(content->rejected); |
| #ifdef WEBRTC_HAVE_SCTP |
| content = |
| cricket::GetFirstDataContent(pc_->local_description()->description()); |
| ASSERT_TRUE(content != NULL); |
| EXPECT_FALSE(content->rejected); |
| #endif |
| } |
| |
| // Test that fallback from DTLS to SDES is not supported. |
| // The fallback was previously supported but was removed to simplify the code |
| // and because it's non-standard. |
| TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| // Wait for fake certificate to be generated. Previously, this is what caused |
| // the "a=crypto" lines to be rejected. |
| AddAudioTrack("audio_label"); |
| AddVideoTrack("video_label"); |
| ASSERT_NE(nullptr, fake_certificate_generator_); |
| EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(), |
| kTimeout); |
| std::unique_ptr<SessionDescriptionInterface> desc( |
| webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, |
| nullptr)); |
| EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false)); |
| } |
| |
| // Test that we can create an audio only offer and receive an answer with a |
| // limited set of audio codecs and receive an updated offer with more audio |
| // codecs, where the added codecs are not supported. |
| TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) { |
| CreatePeerConnectionWithoutDtls(); |
| AddAudioTrack("audio_label"); |
| CreateOfferAsLocalDescription(); |
| |
| const char* answer_sdp = |
| (sdp_semantics_ == SdpSemantics::kPlanB ? webrtc::kAudioSdpPlanB |
| : webrtc::kAudioSdpUnifiedPlan); |
| std::unique_ptr<SessionDescriptionInterface> answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr)); |
| EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false)); |
| |
| const char* reoffer_sdp = |
| (sdp_semantics_ == SdpSemantics::kPlanB |
| ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB |
| : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan); |
| std::unique_ptr<SessionDescriptionInterface> updated_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr)); |
| EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false)); |
| CreateAnswerAsLocalDescription(); |
| } |
| |
| // Test that if we're receiving (but not sending) a track, subsequent offers |
| // will have m-lines with a=recvonly. |
| TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| CreateAndSetRemoteOffer(GetSdpStringWithStream1()); |
| CreateAnswerAsLocalDescription(); |
| |
| // At this point we should be receiving stream 1, but not sending anything. |
| // A new offer should be recvonly. |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| DoCreateOffer(&offer, nullptr); |
| |
| const cricket::ContentInfo* video_content = |
| cricket::GetFirstVideoContent(offer->description()); |
| ASSERT_EQ(RtpTransceiverDirection::kRecvOnly, |
| video_content->media_description()->direction()); |
| |
| const cricket::ContentInfo* audio_content = |
| cricket::GetFirstAudioContent(offer->description()); |
| ASSERT_EQ(RtpTransceiverDirection::kRecvOnly, |
| audio_content->media_description()->direction()); |
| } |
| |
| // Test that if we're receiving (but not sending) a track, and the |
| // offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to |
| // false, the generated m-lines will be a=inactive. |
| TEST_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) { |
| RTCConfiguration rtc_config; |
| rtc_config.enable_dtls_srtp = true; |
| CreatePeerConnection(rtc_config); |
| CreateAndSetRemoteOffer(GetSdpStringWithStream1()); |
| CreateAnswerAsLocalDescription(); |
| |
| // At this point we should be receiving stream 1, but not sending anything. |
| // A new offer would be recvonly, but we'll set the "no receive" constraints |
| // to make it inactive. |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| RTCOfferAnswerOptions options; |
| options.offer_to_receive_audio = 0; |
| options.offer_to_receive_video = 0; |
| DoCreateOffer(&offer, &options); |
| |
| const cricket::ContentInfo* video_content = |
| cricket::GetFirstVideoContent(offer->description()); |
| ASSERT_EQ(RtpTransceiverDirection::kInactive, |
| video_content->media_description()->direction()); |
| |
| const cricket::ContentInfo* audio_content = |
| cricket::GetFirstAudioContent(offer->description()); |
| ASSERT_EQ(RtpTransceiverDirection::kInactive, |
| audio_content->media_description()->direction()); |
| } |
| |
| // Test that we can use SetConfiguration to change the ICE servers of the |
| // PortAllocator. |
| TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) { |
| CreatePeerConnection(); |
| |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| PeerConnectionInterface::IceServer server; |
| server.uri = "stun:test_hostname"; |
| config.servers.push_back(server); |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| |
| EXPECT_EQ(1u, port_allocator_->stun_servers().size()); |
| EXPECT_EQ("test_hostname", |
| port_allocator_->stun_servers().begin()->hostname()); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) { |
| CreatePeerConnection(); |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| config.type = PeerConnectionInterface::kRelay; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter()); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.prune_turn_ports = false; |
| CreatePeerConnection(config); |
| config = pc_->GetConfiguration(); |
| EXPECT_EQ(webrtc::NO_PRUNE, port_allocator_->turn_port_prune_policy()); |
| |
| config.prune_turn_ports = true; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY, |
| port_allocator_->turn_port_prune_policy()); |
| } |
| |
| // Test that the ice check interval can be changed. This does not verify that |
| // the setting makes it all the way to P2PTransportChannel, as that would |
| // require a very complex set of mocks. |
| TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.ice_check_min_interval = absl::nullopt; |
| CreatePeerConnection(config); |
| config = pc_->GetConfiguration(); |
| config.ice_check_min_interval = 100; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| config = pc_->GetConfiguration(); |
| EXPECT_EQ(config.ice_check_min_interval, 100); |
| } |
| |
| TEST_P(PeerConnectionInterfaceTest, |
| SetConfigurationChangesSurfaceIceCandidatesOnIceTransportTypeChanged) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.surface_ice_candidates_on_ice_transport_type_changed = false; |
| CreatePeerConnection(config); |
| config = pc_->GetConfiguration(); |
| EXPECT_FALSE(config.surface_ice_candidates_on_ice_transport_type_changed); |
| |
| config.surface_ice_candidates_on_ice_transport_type_changed = true; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| config = pc_->GetConfiguration(); |
| EXPECT_TRUE(config.surface_ice_candidates_on_ice_transport_type_changed); |
| } |
| |
| // Test that when SetConfiguration changes both the pool size and other |
| // attributes, the pooled session is created with the updated attributes. |
| TEST_P(PeerConnectionInterfaceTest, |
| SetConfigurationCreatesPooledSessionCorrectly) { |
| CreatePeerConnection(); |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| config.ice_candidate_pool_size = 1; |
| PeerConnectionInterface::IceServer server; |
| server.uri = kStunAddressOnly; |
| config.servers.push_back(server); |
| config.type = PeerConnectionInterface::kRelay; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| |
| const cricket::FakePortAllocatorSession* session = |
| static_cast<const cricket::FakePortAllocatorSession*>( |
| port_allocator_->GetPooledSession()); |
| ASSERT_NE(nullptr, session); |
| EXPECT_EQ(1UL, session->stun_servers().size()); |
| } |
| |
| // Test that after SetLocalDescription, changing the pool size is not allowed, |
| // and an invalid modification error is returned. |
| TEST_P(PeerConnectionInterfaceTest, |
| CantChangePoolSizeAfterSetLocalDescription) { |
| CreatePeerConnection(); |
| // Start by setting a size of 1. |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| config.ice_candidate_pool_size = 1; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| |
| // Set remote offer; can still change pool size at this point. |
| CreateOfferAsRemoteDescription(); |
| config.ice_candidate_pool_size = 2; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| |
| // Set local answer; now it's too late. |
| CreateAnswerAsLocalDescription(); |
| config.ice_candidate_pool_size = 3; |
| RTCError error = pc_->SetConfiguration(config); |
| EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type()); |
| } |
| |
| // Test that after setting an answer, extra pooled sessions are discarded. The |
| // ICE candidate pool is only intended to be used for the first offer/answer. |
| TEST_P(PeerConnectionInterfaceTest, |
| ExtraPooledSessionsDiscardedAfterApplyingAnswer) { |
| CreatePeerConnection(); |
| |
| // Set a larger-than-necessary size. |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| config.ice_candidate_pool_size = 4; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| |
| // Do offer/answer. |
| CreateOfferAsRemoteDescription(); |
| CreateAnswerAsLocalDescription(); |
| |
| // Expect no pooled sessions to be left. |
| const cricket::PortAllocatorSession* session = |
| port_allocator_->GetPooledSession(); |
| EXPECT_EQ(nullptr, session); |
| } |
| |
| // After Close is called, pooled candidates should be discarded so as to not |
| // waste network resources. |
| TEST_P(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) { |
| CreatePeerConnection(); |
| |
| PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); |
| config.ice_candidate_pool_size = 3; |
| EXPECT_TRUE(pc_->SetConfiguration(config).ok()); |
| pc_->Close(); |
| |
| // Expect no pooled sessions to be left. |
| const cricket::PortAllocatorSession* session = |
| port_allocator_->GetPooledSession(); |
| EXPECT_EQ(nullptr, session); |
| } |
| |
| // Test that SetConfiguration returns an invalid modification error if |
| // modifying a field in the configuration that isn't allowed to be modified. |
| TEST_P(PeerConnectionInterfaceTest, |
| SetConfigurationReturnsInvalidModificationError) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced; |
| config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate; |
| config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE; |
| CreatePeerConnection(config); |
| |
| PeerConnectionInterface::RTCConfiguration modified_config = |
| pc_->GetConfiguration(); |
| modified_config.bundle_policy = |
| PeerConnectionInterface::kBundlePolicyMaxBundle; |
| RTCError error = pc_->SetConfiguration(modified_config); |
| EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type()); |
| |
| modified_config = pc_->GetConfiguration(); |
| modified_config.rtcp_mux_policy = |
| PeerConnectionInterface::kRtcpMuxPolicyRequire; |
| error = pc_->SetConfiguration(modified_config); |
| EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type()); |
| |
| modified_config = pc_->GetConfiguration(); |
| modified_config.continual_gathering_policy = |
| PeerConnectionInterface::GATHER_CONTINUALLY; |
| error = pc_->SetConfiguration(modified_config); |
| EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type()); |
| } |
| |
| // Test that SetConfiguration returns a range error if the candidate pool size |
| // is negative or larger than allowed by the spec. |
| TEST_P(PeerConnectionInterfaceTest, |
| SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) { |
| PeerConnectionInterface::RTCConfiguration config; |
| CreatePeerConnection(config); |
| config = pc_->GetConfiguration(); |
| |
| config.ice_candidate_pool_size = -1; |
| RTCError error = pc_->SetConfiguration(config); |
| EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type()); |
| |
| config.ice_candidate_pool_size = INT_MAX; |
| error = pc_->SetConfiguration(config); |
| EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type()); |
| } |
| |
| // Test that SetConfiguration returns a syntax error if parsing an ICE server |
| // URL failed. |
| TEST_P(PeerConnectionInterfaceTest, |
| SetConfigurationReturnsSyntaxErrorFromBadIceUrls) { |
| PeerConnectionInterface::RTCConfiguration config; |
| CreatePeerConnection(config); |
| config = pc_->GetConfiguration(); |
| |
| PeerConnectionInterface::IceServer bad_server; |
| bad_server.uri = "stunn:www.example.com"; |
| config.servers.push_back(bad_server); |
| RTCError error = pc_->SetConfiguration(config); |
| EXPECT_EQ(RTCErrorType::SYNTAX_ERROR, error.type()); |
| } |
| |
| // Test that SetConfiguration returns an invalid parameter error if a TURN |
| // IceServer is missing a username or password. |
| TEST_P(PeerConnectionInterfaceTest, |
| SetConfigurationReturnsInvalidParameterIfCredentialsMissing) { |
| PeerConnectionInterface::RTCConfiguration config; |
| CreatePeerConnection(config); |
| config = pc_->GetConfiguration(); |
| |
| PeerConnectionInterface::IceServer bad_server; |
| bad_server.uri = "turn:www.example.com"; |
| // Missing password. |
| bad_server.username = "foo"; |
| config.servers.push_back(bad_server); |
| RTCError error; |
| EXPECT_EQ(pc_->SetConfiguration(config).type(), |
| RTCErrorType::INVALID_PARAMETER); |
| } |
| |
| // Test that PeerConnection::Close changes the states to closed and all remote |
| // tracks change state to ended. |
| TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) { |
| // Initialize a PeerConnection and negotiate local and remote session |
| // description. |
| InitiateCall(); |
| |
| // With Plan B, verify the stream count. The analog with Unified Plan is the |
| // RtpTransceiver count. |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| ASSERT_EQ(1u, pc_->local_streams()->count()); |
| ASSERT_EQ(1u, pc_->remote_streams()->count()); |
| } else { |
| ASSERT_EQ(2u, pc_->GetTransceivers().size()); |
| } |
| |
| pc_->Close(); |
| |
| EXPECT_EQ(PeerConnectionInterface::kClosed, pc_->signaling_state()); |
| EXPECT_EQ(PeerConnectionInterface::kIceConnectionClosed, |
| pc_->ice_connection_state()); |
| EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete, |
| pc_->ice_gathering_state()); |
| |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| EXPECT_EQ(1u, pc_->local_streams()->count()); |
| EXPECT_EQ(1u, pc_->remote_streams()->count()); |
| } else { |
| // Verify that the RtpTransceivers are still returned. |
| EXPECT_EQ(2u, pc_->GetTransceivers().size()); |
| } |
| |
| auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO); |
| auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| ASSERT_TRUE(audio_receiver); |
| ASSERT_TRUE(video_receiver); |
| // Track state may be updated asynchronously. |
| EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, |
| audio_receiver->track()->state(), kTimeout); |
| EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, |
| video_receiver->track()->state(), kTimeout); |
| } else { |
| ASSERT_FALSE(audio_receiver); |
| ASSERT_FALSE(video_receiver); |
| } |
| } |
| |
| // Test that PeerConnection methods fails gracefully after |
| // PeerConnection::Close has been called. |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) { |
| CreatePeerConnectionWithoutDtls(); |
| AddAudioVideoStream(kStreamId1, "audio_label", "video_label"); |
| CreateOfferAsRemoteDescription(); |
| CreateAnswerAsLocalDescription(); |
| |
| ASSERT_EQ(1u, pc_->local_streams()->count()); |
| rtc::scoped_refptr<MediaStreamInterface> local_stream = |
| pc_->local_streams()->at(0); |
| |
| pc_->Close(); |
| |
| pc_->RemoveStream(local_stream); |
| EXPECT_FALSE(pc_->AddStream(local_stream)); |
| |
| EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL); |
| |
| EXPECT_TRUE(pc_->local_description() != NULL); |
| EXPECT_TRUE(pc_->remote_description() != NULL); |
| |
| std::unique_ptr<SessionDescriptionInterface> offer; |
| EXPECT_FALSE(DoCreateOffer(&offer, nullptr)); |
| std::unique_ptr<SessionDescriptionInterface> answer; |
| EXPECT_FALSE(DoCreateAnswer(&answer, nullptr)); |
| |
| std::string sdp; |
| ASSERT_TRUE(pc_->remote_description()->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> remote_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer))); |
| |
| ASSERT_TRUE(pc_->local_description()->ToString(&sdp)); |
| std::unique_ptr<SessionDescriptionInterface> local_offer( |
| webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); |
| EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer))); |
| } |
| |
| // Test that GetStats can still be called after PeerConnection::Close. |
| TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) { |
| InitiateCall(); |
| pc_->Close(); |
| DoGetStats(NULL); |
| } |
| |
| // NOTE: The series of tests below come from what used to be |
| // mediastreamsignaling_unittest.cc, and are mostly aimed at testing that |
| // setting a remote or local description has the expected effects. |
| |
| // This test verifies that the remote MediaStreams corresponding to a received |
| // SDP string is created. In this test the two separate MediaStreams are |
| // signaled. |
| TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
| CreatePeerConnection(config); |
| CreateAndSetRemoteOffer(GetSdpStringWithStream1()); |
| |
| rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1)); |
| EXPECT_TRUE( |
| CompareStreamCollections(observer_.remote_streams(), reference.get())); |
| MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0); |
| EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != nullptr); |
| |
| // Create a session description based on another SDP with another |
| // MediaStream. |
| CreateAndSetRemoteOffer(GetSdpStringWithStream1And2()); |
| |
| rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1)); |
| EXPECT_TRUE( |
| CompareStreamCollections(observer_.remote_streams(), reference2.get())); |
| } |
| |
| // This test verifies that when remote tracks are added/removed from SDP, the |
| // created remote streams are updated appropriately. |
| // Don't run under Unified Plan since this test uses Plan B SDP to test Plan B |
| // specific behavior. |
| TEST_F(PeerConnectionInterfaceTestPlanB, |
| AddRemoveTrackFromExistingRemoteMediaStream) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
| CreatePeerConnection(config); |
| std::unique_ptr<SessionDescriptionInterface> desc_ms1 = |
| CreateSessionDescriptionAndReference(1, 1); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1))); |
| EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(), |
| reference_collection_)); |
| |
| // Add extra audio and video tracks to the same MediaStream. |
| std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks = |
| CreateSessionDescriptionAndReference(2, 2); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1_two_tracks))); |
| EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(), |
| reference_collection_)); |
| rtc::scoped_refptr<AudioTrackInterface> audio_track2 = |
| observer_.remote_streams()->at(0)->GetAudioTracks()[1]; |
| EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state()); |
| rtc::scoped_refptr<VideoTrackInterface> video_track2 = |
| observer_.remote_streams()->at(0)->GetVideoTracks()[1]; |
| EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state()); |
| |
| // Remove the extra audio and video tracks. |
| std::unique_ptr<SessionDescriptionInterface> desc_ms2 = |
| CreateSessionDescriptionAndReference(1, 1); |
| MockTrackObserver audio_track_observer(audio_track2); |
| MockTrackObserver video_track_observer(video_track2); |
| |
| EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1)); |
| EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1)); |
| EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms2))); |
| EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(), |
| reference_collection_)); |
| // Track state may be updated asynchronously. |
| EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded, |
| audio_track2->state(), kTimeout); |
| EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded, |
| video_track2->state(), kTimeout); |
| } |
| |
| // This tests that remote tracks are ended if a local session description is set |
| // that rejects the media content type. |
| TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
| CreatePeerConnection(config); |
| // First create and set a remote offer, then reject its video content in our |
| // answer. |
| CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB); |
| auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO); |
| ASSERT_TRUE(audio_receiver); |
| auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO); |
| ASSERT_TRUE(video_receiver); |
| |
| rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio = |
| audio_receiver->track(); |
| EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state()); |
| rtc::scoped_refptr<MediaStreamTrackInterface> remote_video = |
| video_receiver->track(); |
| EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state()); |
| |
| std::unique_ptr<SessionDescriptionInterface> local_answer; |
| EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr)); |
| cricket::ContentInfo* video_info = |
| local_answer->description()->GetContentByName("video"); |
| video_info->rejected = true; |
| EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer))); |
| EXPECT_EQ(MediaStreamTrackInterface::kEnded, remote_video->state()); |
| EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state()); |
| |
| // Now create an offer where we reject both video and audio. |
| std::unique_ptr<SessionDescriptionInterface> local_offer; |
| EXPECT_TRUE(DoCreateOffer(&local_offer, nullptr)); |
| video_info = local_offer->description()->GetContentByName("video"); |
| ASSERT_TRUE(video_info != nullptr); |
| video_info->rejected = true; |
| cricket::ContentInfo* audio_info = |
| local_offer->description()->GetContentByName("audio"); |
| ASSERT_TRUE(audio_info != nullptr); |
| audio_info->rejected = true; |
| EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer))); |
| // Track state may be updated asynchronously. |
| EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_audio->state(), |
| kTimeout); |
| EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_video->state(), |
| kTimeout); |
| } |
| |
| // This tests that we won't crash if the remote track has been removed outside |
| // of PeerConnection and then PeerConnection tries to reject the track. |
| // Don't run under Unified Plan since the stream API is not available. |
| TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
| CreatePeerConnection(config); |
| CreateAndSetRemoteOffer(GetSdpStringWithStream1()); |
| MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0); |
| remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]); |
| remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]); |
| |
| std::unique_ptr<SessionDescriptionInterface> local_answer( |
| webrtc::CreateSessionDescription(SdpType::kAnswer, |
| GetSdpStringWithStream1(), nullptr)); |
| cricket::ContentInfo* video_info = |
| local_answer->description()->GetContentByName("video"); |
| video_info->rejected = true; |
| cricket::ContentInfo* audio_info = |
| local_answer->description()->GetContentByName("audio"); |
| audio_info->rejected = true; |
| EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer))); |
| |
| // No crash is a pass. |
| } |
| |
| // This tests that if a recvonly remote description is set, no remote streams |
| // will be created, even if the description contains SSRCs/MSIDs. |
| // See: https://code.google.com/p/webrtc/issues/detail?id=5054 |
| TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
| CreatePeerConnection(config); |
| |
| std::string recvonly_offer = GetSdpStringWithStream1(); |
| absl::StrReplaceAll({{kSendrecv, kRecvonly}}, &recvonly_offer); |
| CreateAndSetRemoteOffer(recvonly_offer); |
| |
| EXPECT_EQ(0u, observer_.remote_streams()->count()); |
| } |
| |
| // This tests that a default MediaStream is created if a remote session |
| // description doesn't contain any streams and no MSID support. |
| // It also tests that the default stream is updated if a video m-line is added |
| // in a subsequent session description. |
| // Don't run under Unified Plan since this behavior is Plan B specific. |
| TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
| CreatePeerConnection(config); |
| CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly); |
| |
| ASSERT_EQ(1u, observer_.remote_streams()->count()); |
| MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0); |
| |
| EXPECT_EQ(1u, remote_stream->GetAudioTracks().size()); |
| EXPECT_EQ(0u, remote_stream->GetVideoTracks().size()); |
| EXPECT_EQ("default", remote_stream->id()); |
| |
| CreateAndSetRemoteOffer(kSdpStringWithoutStreams); |
| ASSERT_EQ(1u, observer_.remote_streams()->count()); |
| ASSERT_EQ(1u, remote_stream->GetAudioTracks().size()); |
| EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id()); |
| EXPECT_EQ(MediaStreamTrackInterface::kLive, |
| remote_stream->GetAudioTracks()[0]->state()); |
| ASSERT_EQ(1u, remote_stream->GetVideoTracks().size()); |
| EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id()); |
| EXPECT_EQ(MediaStreamTrackInterface::kLive, |
| remote_stream->GetVideoTracks()[0]->state()); |
| } |
| |
| // This tests that a default MediaStream is created if a remote session |
| // description doesn't contain any streams and media direction is send only. |
| // Don't run under Unified Plan since this behavior is Plan B specific. |
| TEST_F(PeerConnectionInterfaceTestPlanB, |
| SendOnlySdpWithoutMsidCreatesDefaultStream) { |
| RTCConfiguration config; |
| config.enable_dtls_srtp = true; |
|