henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 1 | /* |
| 2 | * libjingle |
jlmiller@webrtc.org | 5f93d0a | 2015-01-20 21:36:13 | [diff] [blame] | 3 | * Copyright 2012 Google Inc. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 4 | * |
| 5 | * Redistribution and use in source and binary forms, with or without |
| 6 | * modification, are permitted provided that the following conditions are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright notice, |
| 9 | * this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright notice, |
| 11 | * this list of conditions and the following disclaimer in the documentation |
| 12 | * and/or other materials provided with the distribution. |
| 13 | * 3. The name of the author may not be used to endorse or promote products |
| 14 | * derived from this software without specific prior written permission. |
| 15 | * |
| 16 | * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| 17 | * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| 18 | * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| 19 | * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| 20 | * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| 21 | * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| 22 | * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| 23 | * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| 24 | * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| 25 | * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 26 | */ |
| 27 | |
| 28 | #ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
| 29 | #define TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
| 30 | |
| 31 | #include <string> |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 32 | #include <vector> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 33 | |
buildbot@webrtc.org | a09a999 | 2014-08-13 17:26:08 | [diff] [blame] | 34 | #include "talk/app/webrtc/datachannel.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 35 | #include "talk/app/webrtc/dtmfsender.h" |
Fredrik Solenberg | 709ed67 | 2015-09-15 10:26:33 | [diff] [blame] | 36 | #include "talk/app/webrtc/mediacontroller.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 37 | #include "talk/app/webrtc/mediastreamprovider.h" |
buildbot@webrtc.org | a09a999 | 2014-08-13 17:26:08 | [diff] [blame] | 38 | #include "talk/app/webrtc/peerconnectioninterface.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 39 | #include "talk/app/webrtc/statstypes.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 40 | #include "talk/media/base/mediachannel.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 41 | #include "talk/session/media/mediasession.h" |
buildbot@webrtc.org | a09a999 | 2014-08-13 17:26:08 | [diff] [blame] | 42 | #include "webrtc/base/sigslot.h" |
Henrik Boström | 5e56c59 | 2015-08-11 08:33:13 | [diff] [blame] | 43 | #include "webrtc/base/sslidentity.h" |
buildbot@webrtc.org | a09a999 | 2014-08-13 17:26:08 | [diff] [blame] | 44 | #include "webrtc/base/thread.h" |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 45 | #include "webrtc/p2p/base/transportcontroller.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 46 | |
| 47 | namespace cricket { |
henrike@webrtc.org | b0ecc1c | 2014-03-26 22:44:28 | [diff] [blame] | 48 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 49 | class ChannelManager; |
| 50 | class DataChannel; |
| 51 | class StatsReport; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 52 | class VideoCapturer; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 53 | class VideoChannel; |
| 54 | class VoiceChannel; |
henrike@webrtc.org | b0ecc1c | 2014-03-26 22:44:28 | [diff] [blame] | 55 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 56 | } // namespace cricket |
| 57 | |
| 58 | namespace webrtc { |
buildbot@webrtc.org | 41451d4 | 2014-05-03 05:39:45 | [diff] [blame] | 59 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 60 | class IceRestartAnswerLatch; |
buildbot@webrtc.org | 41451d4 | 2014-05-03 05:39:45 | [diff] [blame] | 61 | class JsepIceCandidate; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 62 | class MediaStreamSignaling; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 63 | class WebRtcSessionDescriptionFactory; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 64 | |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 65 | extern const char kBundleWithoutRtcpMux[]; |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 66 | extern const char kCreateChannelFailed[]; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 67 | extern const char kInvalidCandidates[]; |
| 68 | extern const char kInvalidSdp[]; |
| 69 | extern const char kMlineMismatch[]; |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 70 | extern const char kPushDownTDFailed[]; |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 | [diff] [blame] | 71 | extern const char kSdpWithoutDtlsFingerprint[]; |
| 72 | extern const char kSdpWithoutSdesCrypto[]; |
mallinath@webrtc.org | 19f27e6 | 2013-10-13 17:18:27 | [diff] [blame] | 73 | extern const char kSdpWithoutIceUfragPwd[]; |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 74 | extern const char kSdpWithoutSdesAndDtlsDisabled[]; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 75 | extern const char kSessionError[]; |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 76 | extern const char kSessionErrorDesc[]; |
pthatcher@webrtc.org | 4eeef58 | 2015-03-16 19:34:23 | [diff] [blame] | 77 | extern const char kDtlsSetupFailureRtp[]; |
| 78 | extern const char kDtlsSetupFailureRtcp[]; |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 79 | extern const char kEnableBundleFailed[]; |
| 80 | |
buildbot@webrtc.org | 53df88c | 2014-08-07 22:46:01 | [diff] [blame] | 81 | // Maximum number of received video streams that will be processed by webrtc |
| 82 | // even if they are not signalled beforehand. |
| 83 | extern const int kMaxUnsignalledRecvStreams; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 84 | |
| 85 | // ICE state callback interface. |
| 86 | class IceObserver { |
| 87 | public: |
wu@webrtc.org | 364f204 | 2013-11-20 21:49:41 | [diff] [blame] | 88 | IceObserver() {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 89 | // Called any time the IceConnectionState changes |
Peter Thatcher | 5436051 | 2015-07-08 18:08:35 | [diff] [blame] | 90 | // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to |
| 91 | // conform to the w3c standard. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 92 | virtual void OnIceConnectionChange( |
| 93 | PeerConnectionInterface::IceConnectionState new_state) {} |
| 94 | // Called any time the IceGatheringState changes |
| 95 | virtual void OnIceGatheringChange( |
| 96 | PeerConnectionInterface::IceGatheringState new_state) {} |
| 97 | // New Ice candidate have been found. |
| 98 | virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0; |
| 99 | // All Ice candidates have been found. |
| 100 | // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange. |
| 101 | // (via PeerConnectionObserver) |
| 102 | virtual void OnIceComplete() {} |
| 103 | |
Peter Thatcher | 5436051 | 2015-07-08 18:08:35 | [diff] [blame] | 104 | // Called whenever the state changes between receiving and not receiving. |
| 105 | virtual void OnIceConnectionReceivingChange(bool receiving) {} |
| 106 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 107 | protected: |
| 108 | ~IceObserver() {} |
wu@webrtc.org | 364f204 | 2013-11-20 21:49:41 | [diff] [blame] | 109 | |
| 110 | private: |
henrikg | 3c089d7 | 2015-09-16 12:37:44 | [diff] [blame] | 111 | RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 112 | }; |
| 113 | |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 114 | // Statistics for all the transports of the session. |
| 115 | typedef std::map<std::string, cricket::TransportStats> TransportStatsMap; |
| 116 | typedef std::map<std::string, std::string> ProxyTransportMap; |
| 117 | |
| 118 | // TODO(pthatcher): Think of a better name for this. We already have |
| 119 | // a TransportStats in transport.h. Perhaps TransportsStats? |
| 120 | struct SessionStats { |
| 121 | ProxyTransportMap proxy_to_transport; |
| 122 | TransportStatsMap transport_stats; |
| 123 | }; |
| 124 | |
| 125 | // A WebRtcSession manages general session state. This includes negotiation |
| 126 | // of both the application-level and network-level protocols: the former |
| 127 | // defines what will be sent and the latter defines how it will be sent. Each |
| 128 | // network-level protocol is represented by a Transport object. Each Transport |
| 129 | // participates in the network-level negotiation. The individual streams of |
| 130 | // packets are represented by TransportChannels. The application-level protocol |
| 131 | // is represented by SessionDecription objects. |
| 132 | class WebRtcSession : public AudioProviderInterface, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 133 | public VideoProviderInterface, |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 | [diff] [blame] | 134 | public DtmfProviderInterface, |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 135 | public DataChannelProviderInterface, |
| 136 | public sigslot::has_slots<> { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 137 | public: |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 138 | enum State { |
| 139 | STATE_INIT = 0, |
| 140 | STATE_SENTOFFER, // Sent offer, waiting for answer. |
| 141 | STATE_RECEIVEDOFFER, // Received an offer. Need to send answer. |
| 142 | STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer. |
| 143 | STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer. |
| 144 | STATE_INPROGRESS, // Offer/answer exchange completed. |
| 145 | STATE_CLOSED, // Close() was called. |
| 146 | }; |
| 147 | |
| 148 | enum Error { |
| 149 | ERROR_NONE = 0, // no error |
| 150 | ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent |
| 151 | ERROR_TRANSPORT = 2, // transport error of some kind |
| 152 | }; |
| 153 | |
stefan | c1aeaf0 | 2015-10-15 14:26:07 | [diff] [blame] | 154 | WebRtcSession(webrtc::MediaControllerInterface* media_controller, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 | [diff] [blame] | 155 | rtc::Thread* signaling_thread, |
| 156 | rtc::Thread* worker_thread, |
deadbeef | ab9b2d1 | 2015-10-14 18:33:11 | [diff] [blame] | 157 | cricket::PortAllocator* port_allocator); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 158 | virtual ~WebRtcSession(); |
| 159 | |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 160 | // These are const to allow them to be called from const methods. |
| 161 | rtc::Thread* signaling_thread() const { return signaling_thread_; } |
| 162 | rtc::Thread* worker_thread() const { return worker_thread_; } |
| 163 | cricket::PortAllocator* port_allocator() const { return port_allocator_; } |
| 164 | |
| 165 | // The ID of this session. |
| 166 | const std::string& id() const { return sid_; } |
| 167 | |
Henrik Lundin | 64dad83 | 2015-05-11 10:44:23 | [diff] [blame] | 168 | bool Initialize( |
| 169 | const PeerConnectionFactoryInterface::Options& options, |
| 170 | const MediaConstraintsInterface* constraints, |
Henrik Boström | 5e56c59 | 2015-08-11 08:33:13 | [diff] [blame] | 171 | rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
Henrik Lundin | 64dad83 | 2015-05-11 10:44:23 | [diff] [blame] | 172 | const PeerConnectionInterface::RTCConfiguration& rtc_configuration); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 173 | // Deletes the voice, video and data channel and changes the session state |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 174 | // to STATE_CLOSED. |
| 175 | void Close(); |
| 176 | |
| 177 | // Returns true if we were the initial offerer. |
| 178 | bool initial_offerer() const { return initial_offerer_; } |
| 179 | |
| 180 | // Returns the current state of the session. See the enum above for details. |
| 181 | // Each time the state changes, we will fire this signal. |
| 182 | State state() const { return state_; } |
| 183 | sigslot::signal2<WebRtcSession*, State> SignalState; |
| 184 | |
| 185 | // Returns the last error in the session. See the enum above for details. |
| 186 | Error error() const { return error_; } |
| 187 | const std::string& error_desc() const { return error_desc_; } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 188 | |
| 189 | void RegisterIceObserver(IceObserver* observer) { |
| 190 | ice_observer_ = observer; |
| 191 | } |
| 192 | |
| 193 | virtual cricket::VoiceChannel* voice_channel() { |
| 194 | return voice_channel_.get(); |
| 195 | } |
| 196 | virtual cricket::VideoChannel* video_channel() { |
| 197 | return video_channel_.get(); |
| 198 | } |
| 199 | virtual cricket::DataChannel* data_channel() { |
| 200 | return data_channel_.get(); |
| 201 | } |
| 202 | |
henrike@webrtc.org | b90991d | 2014-03-04 19:54:57 | [diff] [blame] | 203 | void SetSdesPolicy(cricket::SecurePolicy secure_policy); |
| 204 | cricket::SecurePolicy SdesPolicy() const; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 205 | |
sergeyu@chromium.org | 0be6aa0 | 2013-08-23 23:21:25 | [diff] [blame] | 206 | // Get current ssl role from transport. |
Taylor Brandstetter | f475d365 | 2016-01-08 23:35:57 | [diff] [blame^] | 207 | bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role); |
| 208 | |
| 209 | // Get current SSL role for this channel's transport. |
| 210 | // If |transport| is null, returns false. |
| 211 | bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role); |
sergeyu@chromium.org | 0be6aa0 | 2013-08-23 23:21:25 | [diff] [blame] | 212 | |
jiayl@webrtc.org | b18bf5e | 2014-08-04 18:34:16 | [diff] [blame] | 213 | void CreateOffer( |
| 214 | CreateSessionDescriptionObserver* observer, |
deadbeef | ab9b2d1 | 2015-10-14 18:33:11 | [diff] [blame] | 215 | const PeerConnectionInterface::RTCOfferAnswerOptions& options, |
| 216 | const cricket::MediaSessionOptions& session_options); |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 217 | void CreateAnswer(CreateSessionDescriptionObserver* observer, |
deadbeef | ab9b2d1 | 2015-10-14 18:33:11 | [diff] [blame] | 218 | const MediaConstraintsInterface* constraints, |
| 219 | const cricket::MediaSessionOptions& session_options); |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 | [diff] [blame] | 220 | // The ownership of |desc| will be transferred after this call. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 221 | bool SetLocalDescription(SessionDescriptionInterface* desc, |
| 222 | std::string* err_desc); |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 | [diff] [blame] | 223 | // The ownership of |desc| will be transferred after this call. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 224 | bool SetRemoteDescription(SessionDescriptionInterface* desc, |
| 225 | std::string* err_desc); |
| 226 | bool ProcessIceMessage(const IceCandidateInterface* ice_candidate); |
buildbot@webrtc.org | 41451d4 | 2014-05-03 05:39:45 | [diff] [blame] | 227 | |
mallinath@webrtc.org | 3d81b1b | 2014-09-09 14:38:10 | [diff] [blame] | 228 | bool SetIceTransports(PeerConnectionInterface::IceTransportsType type); |
buildbot@webrtc.org | 41451d4 | 2014-05-03 05:39:45 | [diff] [blame] | 229 | |
honghaiz | 1f429e3 | 2015-09-28 14:57:34 | [diff] [blame] | 230 | cricket::IceConfig ParseIceConfig( |
| 231 | const PeerConnectionInterface::RTCConfiguration& config) const; |
| 232 | |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 233 | void SetIceConfig(const cricket::IceConfig& ice_config); |
| 234 | |
| 235 | // Start gathering candidates for any new transports, or transports doing an |
| 236 | // ICE restart. |
| 237 | void MaybeStartGathering(); |
| 238 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 239 | const SessionDescriptionInterface* local_description() const { |
| 240 | return local_desc_.get(); |
| 241 | } |
| 242 | const SessionDescriptionInterface* remote_description() const { |
| 243 | return remote_desc_.get(); |
| 244 | } |
| 245 | |
| 246 | // Get the id used as a media stream track's "id" field from ssrc. |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 247 | virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id); |
| 248 | virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id); |
xians@webrtc.org | 4cb0128 | 2014-06-12 14:57:05 | [diff] [blame] | 249 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 250 | // AudioMediaProviderInterface implementation. |
solenberg | d4cec0d | 2015-10-09 15:55:48 | [diff] [blame] | 251 | void SetAudioPlayout(uint32_t ssrc, bool enable) override; |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 252 | void SetAudioSend(uint32_t ssrc, |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 | [diff] [blame] | 253 | bool enable, |
| 254 | const cricket::AudioOptions& options, |
| 255 | cricket::AudioRenderer* renderer) override; |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 256 | void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override; |
Tommi | f888bb5 | 2015-12-12 00:37:01 | [diff] [blame] | 257 | void SetRawAudioSink(uint32_t ssrc, |
| 258 | rtc::scoped_ptr<AudioSinkInterface> sink) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 259 | |
| 260 | // Implements VideoMediaProviderInterface. |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 261 | bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override; |
| 262 | void SetVideoPlayout(uint32_t ssrc, |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 | [diff] [blame] | 263 | bool enable, |
| 264 | cricket::VideoRenderer* renderer) override; |
Peter Boström | 0c4e06b | 2015-10-07 10:23:21 | [diff] [blame] | 265 | void SetVideoSend(uint32_t ssrc, |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 | [diff] [blame] | 266 | bool enable, |
| 267 | const cricket::VideoOptions* options) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 268 | |
| 269 | // Implements DtmfProviderInterface. |
| 270 | virtual bool CanInsertDtmf(const std::string& track_id); |
| 271 | virtual bool InsertDtmf(const std::string& track_id, |
| 272 | int code, int duration); |
| 273 | virtual sigslot::signal0<>* GetOnDestroyedSignal(); |
| 274 | |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 | [diff] [blame] | 275 | // Implements DataChannelProviderInterface. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 | [diff] [blame] | 276 | bool SendData(const cricket::SendDataParams& params, |
| 277 | const rtc::Buffer& payload, |
| 278 | cricket::SendDataResult* result) override; |
| 279 | bool ConnectDataChannel(DataChannel* webrtc_data_channel) override; |
| 280 | void DisconnectDataChannel(DataChannel* webrtc_data_channel) override; |
| 281 | void AddSctpDataStream(int sid) override; |
| 282 | void RemoveSctpDataStream(int sid) override; |
| 283 | bool ReadyToSendData() const override; |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 | [diff] [blame] | 284 | |
pthatcher@webrtc.org | c04a97f | 2015-03-16 19:31:40 | [diff] [blame] | 285 | // Returns stats for all channels of all transports. |
| 286 | // This avoids exposing the internal structures used to track them. |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 287 | virtual bool GetTransportStats(SessionStats* stats); |
pthatcher@webrtc.org | c04a97f | 2015-03-16 19:31:40 | [diff] [blame] | 288 | |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 289 | // Get stats for a specific channel |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 290 | bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats); |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 291 | |
| 292 | // virtual so it can be mocked in unit tests |
| 293 | virtual bool GetLocalCertificate( |
| 294 | const std::string& transport_name, |
| 295 | rtc::scoped_refptr<rtc::RTCCertificate>* certificate); |
| 296 | |
| 297 | // Caller owns returned certificate |
| 298 | virtual bool GetRemoteSSLCertificate(const std::string& transport_name, |
| 299 | rtc::SSLCertificate** cert); |
| 300 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 301 | cricket::DataChannelType data_channel_type() const; |
| 302 | |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 303 | bool IceRestartPending() const; |
| 304 | |
| 305 | void ResetIceRestartLatch(); |
| 306 | |
Henrik Boström | d828198 | 2015-08-27 08:12:24 | [diff] [blame] | 307 | // Called when an RTCCertificate is generated or retrieved by |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 308 | // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |
Henrik Boström | d828198 | 2015-08-27 08:12:24 | [diff] [blame] | 309 | void OnCertificateReady( |
| 310 | const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
pthatcher@webrtc.org | 4eeef58 | 2015-03-16 19:34:23 | [diff] [blame] | 311 | void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 312 | |
| 313 | // For unit test. |
Henrik Boström | d828198 | 2015-08-27 08:12:24 | [diff] [blame] | 314 | bool waiting_for_certificate_for_testing() const; |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 315 | const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 316 | |
guoweis@webrtc.org | 7169afd | 2014-12-04 17:59:29 | [diff] [blame] | 317 | void set_metrics_observer( |
| 318 | webrtc::MetricsObserverInterface* metrics_observer) { |
| 319 | metrics_observer_ = metrics_observer; |
| 320 | } |
| 321 | |
deadbeef | ab9b2d1 | 2015-10-14 18:33:11 | [diff] [blame] | 322 | // Called when voice_channel_, video_channel_ and data_channel_ are created |
| 323 | // and destroyed. As a result of, for example, setting a new description. |
| 324 | sigslot::signal0<> SignalVoiceChannelCreated; |
| 325 | sigslot::signal0<> SignalVoiceChannelDestroyed; |
| 326 | sigslot::signal0<> SignalVideoChannelCreated; |
| 327 | sigslot::signal0<> SignalVideoChannelDestroyed; |
| 328 | sigslot::signal0<> SignalDataChannelCreated; |
| 329 | sigslot::signal0<> SignalDataChannelDestroyed; |
| 330 | |
| 331 | // Called when a valid data channel OPEN message is received. |
| 332 | // std::string represents the data channel label. |
| 333 | sigslot::signal2<const std::string&, const InternalDataChannelInit&> |
| 334 | SignalDataChannelOpenMessage; |
| 335 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 336 | private: |
| 337 | // Indicates the type of SessionDescription in a call to SetLocalDescription |
| 338 | // and SetRemoteDescription. |
| 339 | enum Action { |
| 340 | kOffer, |
| 341 | kPrAnswer, |
| 342 | kAnswer, |
| 343 | }; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 344 | |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 345 | // Log session state. |
| 346 | void LogState(State old_state, State new_state); |
| 347 | |
| 348 | // Updates the state, signaling if necessary. |
| 349 | virtual void SetState(State state); |
| 350 | |
| 351 | // Updates the error state, signaling if necessary. |
| 352 | // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|. |
| 353 | virtual void SetError(Error error, const std::string& error_desc); |
| 354 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 355 | bool UpdateSessionState(Action action, cricket::ContentSource source, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 356 | std::string* err_desc); |
| 357 | static Action GetAction(const std::string& type); |
pthatcher@webrtc.org | 592470b | 2015-03-16 21:15:37 | [diff] [blame] | 358 | // Push the media parts of the local or remote session description |
| 359 | // down to all of the channels. |
| 360 | bool PushdownMediaDescription(cricket::ContentAction action, |
| 361 | cricket::ContentSource source, |
| 362 | std::string* error_desc); |
| 363 | |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 364 | bool PushdownTransportDescription(cricket::ContentSource source, |
| 365 | cricket::ContentAction action, |
| 366 | std::string* error_desc); |
| 367 | |
| 368 | // Helper methods to push local and remote transport descriptions. |
| 369 | bool PushdownLocalTransportDescription( |
| 370 | const cricket::SessionDescription* sdesc, |
| 371 | cricket::ContentAction action, |
| 372 | std::string* error_desc); |
| 373 | bool PushdownRemoteTransportDescription( |
| 374 | const cricket::SessionDescription* sdesc, |
| 375 | cricket::ContentAction action, |
| 376 | std::string* error_desc); |
| 377 | |
| 378 | // Returns true and the TransportInfo of the given |content_name| |
| 379 | // from |description|. Returns false if it's not available. |
| 380 | static bool GetTransportDescription( |
| 381 | const cricket::SessionDescription* description, |
| 382 | const std::string& content_name, |
| 383 | cricket::TransportDescription* info); |
| 384 | |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 385 | cricket::BaseChannel* GetChannel(const std::string& content_name); |
| 386 | // Cause all the BaseChannels in the bundle group to have the same |
| 387 | // transport channel. |
| 388 | bool EnableBundle(const cricket::ContentGroup& bundle); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 389 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 390 | // Enables media channels to allow sending of media. |
| 391 | void EnableChannels(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 392 | // Returns the media index for a local ice candidate given the content name. |
| 393 | // Returns false if the local session description does not have a media |
| 394 | // content called |content_name|. |
| 395 | bool GetLocalCandidateMediaIndex(const std::string& content_name, |
| 396 | int* sdp_mline_index); |
| 397 | // Uses all remote candidates in |remote_desc| in this session. |
| 398 | bool UseCandidatesInSessionDescription( |
| 399 | const SessionDescriptionInterface* remote_desc); |
| 400 | // Uses |candidate| in this session. |
| 401 | bool UseCandidate(const IceCandidateInterface* candidate); |
| 402 | // Deletes the corresponding channel of contents that don't exist in |desc|. |
| 403 | // |desc| can be null. This means that all channels are deleted. |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 404 | void RemoveUnusedChannels(const cricket::SessionDescription* desc); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 405 | |
| 406 | // Allocates media channels based on the |desc|. If |desc| doesn't have |
| 407 | // the BUNDLE option, this method will disable BUNDLE in PortAllocator. |
| 408 | // This method will also delete any existing media channels before creating. |
| 409 | bool CreateChannels(const cricket::SessionDescription* desc); |
| 410 | |
| 411 | // Helper methods to create media channels. |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 412 | bool CreateVoiceChannel(const cricket::ContentInfo* content); |
| 413 | bool CreateVideoChannel(const cricket::ContentInfo* content); |
| 414 | bool CreateDataChannel(const cricket::ContentInfo* content); |
| 415 | |
henrika@webrtc.org | aebb1ad | 2014-01-14 10:00:58 | [diff] [blame] | 416 | // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN |
| 417 | // messages. |
| 418 | void OnDataChannelMessageReceived(cricket::DataChannel* channel, |
| 419 | const cricket::ReceiveDataParams& params, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 | [diff] [blame] | 420 | const rtc::Buffer& payload); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 421 | |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 422 | std::string BadStateErrMsg(State state); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 423 | void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state); |
Peter Thatcher | 5436051 | 2015-07-08 18:08:35 | [diff] [blame] | 424 | void SetIceConnectionReceiving(bool receiving); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 425 | |
sergeyu@chromium.org | 0be6aa0 | 2013-08-23 23:21:25 | [diff] [blame] | 426 | bool ValidateBundleSettings(const cricket::SessionDescription* desc); |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 427 | bool HasRtcpMuxEnabled(const cricket::ContentInfo* content); |
sergeyu@chromium.org | 0be6aa0 | 2013-08-23 23:21:25 | [diff] [blame] | 428 | // Below methods are helper methods which verifies SDP. |
| 429 | bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc, |
| 430 | cricket::ContentSource source, |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 431 | std::string* err_desc); |
sergeyu@chromium.org | 0be6aa0 | 2013-08-23 23:21:25 | [diff] [blame] | 432 | |
| 433 | // Check if a call to SetLocalDescription is acceptable with |action|. |
| 434 | bool ExpectSetLocalDescription(Action action); |
| 435 | // Check if a call to SetRemoteDescription is acceptable with |action|. |
| 436 | bool ExpectSetRemoteDescription(Action action); |
| 437 | // Verifies a=setup attribute as per RFC 5763. |
| 438 | bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc, |
| 439 | Action action); |
henrike@webrtc.org | 1e09a71 | 2013-07-26 19:17:59 | [diff] [blame] | 440 | |
jiayl@webrtc.org | e10d28c | 2014-07-17 17:07:49 | [diff] [blame] | 441 | // Returns true if we are ready to push down the remote candidate. |
| 442 | // |remote_desc| is the new remote description, or NULL if the current remote |
| 443 | // description should be used. Output |valid| is true if the candidate media |
| 444 | // index is valid. |
| 445 | bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate, |
| 446 | const SessionDescriptionInterface* remote_desc, |
| 447 | bool* valid); |
| 448 | |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 449 | void OnTransportControllerConnectionState(cricket::IceConnectionState state); |
| 450 | void OnTransportControllerReceiving(bool receiving); |
| 451 | void OnTransportControllerGatheringState(cricket::IceGatheringState state); |
| 452 | void OnTransportControllerCandidatesGathered( |
| 453 | const std::string& transport_name, |
| 454 | const cricket::Candidates& candidates); |
| 455 | |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 | [diff] [blame] | 456 | std::string GetSessionErrorMsg(); |
| 457 | |
deadbeef | cbecd35 | 2015-09-23 18:50:27 | [diff] [blame] | 458 | // Invoked when TransportController connection completion is signaled. |
| 459 | // Reports stats for all transports in use. |
| 460 | void ReportTransportStats(); |
| 461 | |
| 462 | // Gather the usage of IPv4/IPv6 as best connection. |
jbauch | ac8869e | 2015-07-03 08:36:14 | [diff] [blame] | 463 | void ReportBestConnectionState(const cricket::TransportStats& stats); |
| 464 | |
| 465 | void ReportNegotiatedCiphers(const cricket::TransportStats& stats); |
guoweis@webrtc.org | 7169afd | 2014-12-04 17:59:29 | [diff] [blame] | 466 | |
stefan | c1aeaf0 | 2015-10-15 14:26:07 | [diff] [blame] | 467 | void OnSentPacket_w(cricket::TransportChannel* channel, |
| 468 | const rtc::SentPacket& sent_packet); |
| 469 | |
deadbeef | d59daf8 | 2015-10-14 22:02:44 | [diff] [blame] | 470 | rtc::Thread* const signaling_thread_; |
| 471 | rtc::Thread* const worker_thread_; |
| 472 | cricket::PortAllocator* const port_allocator_; |
| 473 | |
| 474 | State state_ = STATE_INIT; |
| 475 | Error error_ = ERROR_NONE; |
| 476 | std::string error_desc_; |
| 477 | |
| 478 | const std::string sid_; |
| 479 | bool initial_offerer_ = false; |
| 480 | |
| 481 | rtc::scoped_ptr<cricket::TransportController> transport_controller_; |
stefan | c1aeaf0 | 2015-10-15 14:26:07 | [diff] [blame] | 482 | MediaControllerInterface* media_controller_; |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 | [diff] [blame] | 483 | rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_; |
| 484 | rtc::scoped_ptr<cricket::VideoChannel> video_channel_; |
| 485 | rtc::scoped_ptr<cricket::DataChannel> data_channel_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 486 | cricket::ChannelManager* channel_manager_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 487 | IceObserver* ice_observer_; |
| 488 | PeerConnectionInterface::IceConnectionState ice_connection_state_; |
Peter Thatcher | 5436051 | 2015-07-08 18:08:35 | [diff] [blame] | 489 | bool ice_connection_receiving_; |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 | [diff] [blame] | 490 | rtc::scoped_ptr<SessionDescriptionInterface> local_desc_; |
| 491 | rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 492 | // If the remote peer is using a older version of implementation. |
| 493 | bool older_version_remote_peer_; |
mallinath@webrtc.org | a27be8e | 2013-09-27 23:04:10 | [diff] [blame] | 494 | bool dtls_enabled_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 495 | // Specifies which kind of data channel is allowed. This is controlled |
| 496 | // by the chrome command-line flag and constraints: |
| 497 | // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled, |
| 498 | // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is |
| 499 | // not set or false, SCTP is allowed (DCT_SCTP); |
| 500 | // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP); |
| 501 | // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE). |
| 502 | cricket::DataChannelType data_channel_type_; |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 | [diff] [blame] | 503 | rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_; |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 504 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 | [diff] [blame] | 505 | rtc::scoped_ptr<WebRtcSessionDescriptionFactory> |
wu@webrtc.org | 91053e7 | 2013-08-10 07:18:04 | [diff] [blame] | 506 | webrtc_session_desc_factory_; |
| 507 | |
henrike@webrtc.org | 6e3dbc2 | 2014-03-25 17:09:47 | [diff] [blame] | 508 | // Member variables for caching global options. |
| 509 | cricket::AudioOptions audio_options_; |
| 510 | cricket::VideoOptions video_options_; |
guoweis@webrtc.org | 7169afd | 2014-12-04 17:59:29 | [diff] [blame] | 511 | MetricsObserverInterface* metrics_observer_; |
henrike@webrtc.org | 6e3dbc2 | 2014-03-25 17:09:47 | [diff] [blame] | 512 | |
pthatcher@webrtc.org | 877ac76 | 2015-02-04 22:03:09 | [diff] [blame] | 513 | // Declares the bundle policy for the WebRTCSession. |
| 514 | PeerConnectionInterface::BundlePolicy bundle_policy_; |
| 515 | |
Peter Thatcher | af55ccc | 2015-05-21 14:48:41 | [diff] [blame] | 516 | // Declares the RTCP mux policy for the WebRTCSession. |
| 517 | PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; |
| 518 | |
henrikg | 3c089d7 | 2015-09-16 12:37:44 | [diff] [blame] | 519 | RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |
wu@webrtc.org | 364f204 | 2013-11-20 21:49:41 | [diff] [blame] | 520 | }; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 | [diff] [blame] | 521 | } // namespace webrtc |
| 522 | |
| 523 | #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ |