WebRtc 之P2C的建立

概述

WebRtc信令交换的过程实际上是基于JSEP01(Javascript Session Establishment Protocol)。
在上一篇[WebRtc建立P2P链接的总体流程]中只是描述了一种简单的形式，建立链接主要需要经过如下过程：

以此为基础总结下p2p建立的过程！

CreatOffer

peerconnection.cc

void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,                                 const MediaConstraintsInterface* constraints) {  if (!VERIFY(observer != NULL)) {    LOG(LS_ERROR) << "CreateOffer - observer is NULL.";    return;  }  RTCOfferAnswerOptions options;  bool value;  size_t mandatory_constraints = 0;  //根据应用层传入的MediaConstraints设置RTCOfferAnswerOptions  if (FindConstraint(constraints,                     MediaConstraintsInterface::kOfferToReceiveAudio,                     &value,                     &mandatory_constraints)) {    options.offer_to_receive_audio =        value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;  }  ......  CreateOffer(observer, options);}void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,                                 const RTCOfferAnswerOptions& options) {  if (!VERIFY(observer != NULL)) {    LOG(LS_ERROR) << "CreateOffer - observer is NULL.";    return;  }  //此处的session为PeerConnection初始化时创建的WebRtcSession对象  session_->CreateOffer(observer, options);}

接下来看WebRtcSession中的CreateOffer是如何实现的，见webrtcsession.cc中：

void WebRtcSession::CreateOffer(    CreateSessionDescriptionObserver* observer,    const PeerConnectionInterface::RTCOfferAnswerOptions& options) {    //webrtc_session_desc_factory_是WebRtcSession在Initialize创建    //根据需求创建是否DTLS加密的WebRtcSessionDescriptionFactory  webrtc_session_desc_factory_->CreateOffer(observer, options);}void WebRtcSessionDescriptionFactory::CreateOffer(    CreateSessionDescriptionObserver* observer,    const PeerConnectionInterface::RTCOfferAnswerOptions& options) {  cricket::MediaSessionOptions session_options;  std::string error = "CreateOffer";  if (certificate_request_state_ == CERTIFICATE_FAILED) {    error += kFailedDueToIdentityFailed;    LOG(LS_ERROR) << error;    PostCreateSessionDescriptionFailed(observer, error);    return;  }/*在Java层org/appspot/apprtc/PeerConnectionClient.java中  private void createPeerConnectionInternal(EGLContext renderEGLContext){  .....      peerConnection = factory.createPeerConnection(        rtcConfig, pcConstraints, pcObserver);    isInitiator = false;.....  mediaStream.addTrack(createVideoTrack(videoCapturer));  ....      mediaStream.addTrack(factory.createAudioTrack(        AUDIO_TRACK_ID,        factory.createAudioSource(audioConstraints)));    peerConnection.addStream(mediaStream);   .....  }  最终是将会话中需要的MediaStream传入mediastream_signaling_进行管理！  所以在mediastream_signaling_中可以获取MediaSessionOptions*/  if (!mediastream_signaling_->GetOptionsForOffer(options,                                                  &session_options)) {    error += " called with invalid options.";    LOG(LS_ERROR) << error;    PostCreateSessionDescriptionFailed(observer, error);    return;  }//检测提供的audio video流是否合法即不能有相同的id  if (!ValidStreams(session_options.streams)) {    error += " called with invalid media streams.";    LOG(LS_ERROR) << error;    PostCreateSessionDescriptionFailed(observer, error);    return;  }  if (data_channel_type_ == cricket::DCT_SCTP &&      mediastream_signaling_->HasDataChannels()) {       //若在应用层建立了datachannel传输用户数据，设置成SCTP协议传输    session_options.data_channel_type = cricket::DCT_SCTP;  }  CreateSessionDescriptionRequest request(      CreateSessionDescriptionRequest::kOffer, observer, session_options);  if (certificate_request_state_ == CERTIFICATE_WAITING) {    create_session_description_requests_.push(request);  } else {    ASSERT(certificate_request_state_ == CERTIFICATE_SUCCEEDED ||           certificate_request_state_ == CERTIFICATE_NOT_NEEDED);    InternalCreateOffer(request); //根据request创建SDP  }}//从上可以看出mediastreamsignaling.cc实际上是peerconnection.cc和webrtcsession.cc沟通的桥梁mediastreamsignaling.cc负责多媒体相关的管理！//根据request创建SDPvoid WebRtcSessionDescriptionFactory::InternalCreateOffer(    CreateSessionDescriptionRequest request) {  cricket::SessionDescription* desc(      session_desc_factory_.CreateOffer(          request.options,          static_cast<cricket::BaseSession*>(session_)->local_description()));  // RFC 3264  // When issuing an offer that modifies the session,  // the "o=" line of the new SDP MUST be identical to that in the  // previous SDP, except that the version in the origin field MUST  // increment by one from the previous SDP.  // Just increase the version number by one each time when a new offer  // is created regardless if it's identical to the previous one or not.  // The |session_version_| is a uint64, the wrap around should not happen.  ASSERT(session_version_ + 1 > session_version_);  //根据JSEP规范创建JsepSessionDescription  JsepSessionDescription* offer(new JsepSessionDescription(      JsepSessionDescription::kOffer));  if (!offer->Initialize(desc, session_id_,                         rtc::ToString(session_version_++))) {    delete offer;    PostCreateSessionDescriptionFailed(request.observer,                                       "Failed to initialize the offer.");    return;  }  if (session_->local_description() &&      !request.options.transport_options.ice_restart) {    // Include all local ice candidates in the SessionDescription unless    // the an ice restart has been requested.    CopyCandidatesFromSessionDescription(session_->local_description(), offer);  }   //将创建的JsepSessionDescription回调给上层的应用，应用层可以结合自己的情况做相应更改  PostCreateSessionDescriptionSucceeded(request.observer, offer); }/*onCreateSuccess为org/appspot/apprtc/PeerConnectionClient.java中offer创建成功后调用的回调，在此会调用根据实际情况修改了音视频编解码相关的信息private class SDPObserver implements SdpObserver {    @Override    public void onCreateSuccess(final SessionDescription origSdp) {      if (localSdp != null) {        reportError("Multiple SDP create.");        return;      }      String sdpDescription = origSdp.description;      if (preferIsac) {        sdpDescription = preferCodec(sdpDescription, AUDIO_CODEC_ISAC, true);      }      if (videoCallEnabled && preferH264) {        sdpDescription = preferCodec(sdpDescription, VIDEO_CODEC_H264, false);      }      final SessionDescription sdp = new SessionDescription(          origSdp.type, sdpDescription);      localSdp = sdp;      executor.execute(new Runnable() {        @Override        public void run() {          if (peerConnection != null && !isError) {            Log.d(TAG, "Set local SDP from " + sdp.type);            peerConnection.setLocalDescription(sdpObserver, sdp);          }        }      });    }*/

SetlocalSDP

下面我们看看PeerConnectionClient.java中回调的实现

private class SDPObserver implements SdpObserver {    @Override    public void onCreateSuccess(final SessionDescription origSdp) {      if (localSdp != null) {        reportError("Multiple SDP create.");        return;      }      String sdpDescription = origSdp.description;      //根据平台，更改音视频的编码方式      if (preferIsac) {        sdpDescription = preferCodec(sdpDescription, AUDIO_CODEC_ISAC, true);      }      if (videoCallEnabled && preferH264) {        sdpDescription = preferCodec(sdpDescription, VIDEO_CODEC_H264, false);      }      //创建最终的SessionDescription      final SessionDescription sdp = new SessionDescription(          origSdp.type, sdpDescription);      localSdp = sdp;      executor.execute(new Runnable() {        @Override        public void run() {          if (peerConnection != null && !isError) {            Log.d(TAG, "Set local SDP from " + sdp.type);            //更新本地的SessionDescription,若设置成功将回调SDPObserver的onSetSuccess方法            peerConnection.setLocalDescription(sdpObserver, sdp);          }        }      });    }    @Override    public void onSetSuccess() {      executor.execute(new Runnable() {        @Override        public void run() {          if (peerConnection == null || isError) {            return;          }          if (isInitiator) {            //发起呼叫端逻辑            // For offering peer connection we first create offer and set            // local SDP, then after receiving answer set remote SDP.            if (peerConnection.getRemoteDescription() == null) {              // We've just set our local SDP so time to send it.              Log.d(TAG, "Local SDP set succesfully");              //将local SDP发送到服务器              events.onLocalDescription(localSdp);            } else {              // We've just set remote description, so drain remote              // and send local ICE candidates.              Log.d(TAG, "Remote SDP set succesfully");              drainCandidates();            }          } else {          //被动应答端逻辑            // For answering peer connection we set remote SDP and then            // create answer and set local SDP.            if (peerConnection.getLocalDescription() != null) {              // We've just set our local SDP so time to send it, drain              // remote and send local ICE candidates.              Log.d(TAG, "Local SDP set succesfully");              events.onLocalDescription(localSdp);              drainCandidates();            } else {              // We've just set remote SDP - do nothing for now -              // answer will be created soon.              Log.d(TAG, "Remote SDP set succesfully");            }          }        }      });    }    .....}

peerConnection.setLocalDescription(sdpObserver, sdp);的实现如下：

void PeerConnection::SetLocalDescription(    SetSessionDescriptionObserver* observer,    SessionDescriptionInterface* desc) {    ......  //检测传入的desc是否合法更新状态后，设置到  if (!session_->SetLocalDescription(desc, &error)) {    PostSetSessionDescriptionFailure(observer, error);    return;  }  SetSessionDescriptionMsg* msg =  new SetSessionDescriptionMsg(observer);   //回调sdpObserver.onSetSuccess将local SDP发送给服务器  signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);  // MaybeStartGathering needs to be called after posting  // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates  // before signaling that SetLocalDescription completed.  //设置LocalDescription后向ICE服务器发出请求StartGathering  session_->MaybeStartGathering(); }bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,                                        std::string* err_desc) {  ASSERT(signaling_thread()->IsCurrent());  // Takes the ownership of |desc| regardless of the result.  rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);  // Validate SDP.  if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {    return false;  }  // Update the initiator flag if this session is the initiator.  Action action = GetAction(desc->type());  if (state() == STATE_INIT && action == kOffer) {    set_initiator(true);  }  .....  //设置本地description  set_local_description(desc->description()->Copy());  local_desc_.reset(desc_temp.release());  // Transport and Media channels will be created only when offer is set.  //CreateChannels根据本地的description创建audiochannel videochannel datachannel  if (action == kOffer && !CreateChannels(local_desc_->description())) {    // TODO(mallinath) - Handle CreateChannel failure, as new local description    // is applied. Restore back to old description.    return BadLocalSdp(desc->type(), kCreateChannelFailed, err_desc);  }    ......  if (remote_description()) {  //如果会话已经设置了远程description    // Now that we have a local description, we can push down remote candidates    // that we stored, and those from the remote description.    if (!saved_candidates_.empty()) {      // If there are saved candidates which arrived before the local      // description was set, copy those to the remote description.      CopySavedCandidates(remote_desc_.get());    }    // Push remote candidates in remote description to transport channels.    //从远程SDP中获取condidate并通过    //TransportController::AddRemoteCandidates-->P2PTransportChannel::AddRemoteCandidate--->bool P2PTransportChannel::CreateConnections    //最终会建立P2P的链接，前面文章提到在ClientB链接到服务器时，服务器会将ClientA的SDP返回给B端，B端首先会设置remote_description    //所以在ClientB设置local description时开始与ClientA建立p2p链接    UseCandidatesInSessionDescription(remote_desc_.get());  }  // Update state and SSRC of local MediaStreams and DataChannels based on the  // local session description.  mediastream_signaling_->OnLocalDescriptionChanged(local_desc_.get());  rtc::SSLRole role;  if (data_channel_type_ == cricket::DCT_SCTP && GetSslRole(&role)) {    mediastream_signaling_->OnDtlsRoleReadyForSctp(role);  }  if (error() != cricket::BaseSession::ERROR_NONE) {    return BadLocalSdp(desc->type(), GetSessionErrorMsg(), err_desc);  }  return true;}

GatheringICECondidate

上面的分析提到，在void PeerConnection::SetLocalDescription中会调用 session_->MaybeStartGathering()开始访问iceserver获取本地的condidate

void BaseSession::MaybeStartGathering() {//直接调用TransportController类中的MaybeStartGathering方法  transport_controller_->MaybeStartGathering();}void TransportController::MaybeStartGathering() {    //在工作线程worker_thread_中调用MaybeStartGathering_w方法！  worker_thread_->Invoke<void>(      rtc::Bind(&TransportController::MaybeStartGathering_w, this));}void TransportController::MaybeStartGathering_w() {//transports_为map<std::string, Transport*> TransportMap，最终会调用P2PTransportChannel::MaybeStartGathering()  for (const auto& kv : transports_) {    kv.second->MaybeStartGathering();  }}void P2PTransportChannel::MaybeStartGathering() {  // Start gathering if we never started before, or if an ICE restart occurred.  if (allocator_sessions_.empty() ||      IceCredentialsChanged(allocator_sessions_.back()->ice_ufrag(),                            allocator_sessions_.back()->ice_pwd(), ice_ufrag_,                            ice_pwd_)) {    if (gathering_state_ != kIceGatheringGathering) {      gathering_state_ = kIceGatheringGathering;      SignalGatheringState(this);    }    // Time for a new allocator    AddAllocatorSession(allocator_->CreateSession(        SessionId(), transport_name(), component(), ice_ufrag_, ice_pwd_));  }}void P2PTransportChannel::AddAllocatorSession(PortAllocatorSession* session) {  session->set_generation(static_cast<uint32>(allocator_sessions_.size()));  allocator_sessions_.push_back(session);  // We now only want to apply new candidates that we receive to the ports  // created by this new session because these are replacing those of the  // previous sessions.  ports_.clear();//通过信号与槽的方式获取底层的通知事件，此处的port不仅仅是传统意义上的端口//实际代表的是一种通信协议，eg:TCPPort ,UDPPort,StunPort,RelayPort  session->SignalPortReady.connect(this, &P2PTransportChannel::OnPortReady);   //底层每发现一个condiatate都会通知到P2PTransportChannel::OnCandidatesReady  session->SignalCandidatesReady.connect(      this, &P2PTransportChannel::OnCandidatesReady);  session->SignalCandidatesAllocationDone.connect(      this, &P2PTransportChannel::OnCandidatesAllocationDone);    // session实际为BasicPortAllocatorSession  session->StartGettingPorts();}void BasicPortAllocatorSession::StartGettingPorts() {  network_thread_ = rtc::Thread::Current();  if (!socket_factory_) {    owned_socket_factory_.reset(        new rtc::BasicPacketSocketFactory(network_thread_));    socket_factory_ = owned_socket_factory_.get();  }  running_ = true;  network_thread_->Post(this, MSG_CONFIG_START);  if (flags() & PORTALLOCATOR_ENABLE_SHAKER)    network_thread_->PostDelayed(ShakeDelay(), this, MSG_SHAKE);}//经过一次处理MSG_CONFIG_START ==>MSG_CONFIG_READY==>MSG_ALLOCATE 消息==>OnAllocate==>DoAllocate()//为本机每一个物理网络分配portsvoid BasicPortAllocatorSession::DoAllocate() {  bool done_signal_needed = false;  std::vector<rtc::Network*> networks;  //获取本机的网络信息eg：ip 。。。  GetNetworks(&networks);  if (networks.empty()) {    LOG(LS_WARNING) << "Machine has no networks; no ports will be allocated";    done_signal_needed = true;  } else {    for (uint32 i = 0; i < networks.size(); ++i) {      PortConfiguration* config = NULL;      if (configs_.size() > 0)        config = configs_.back();      uint32 sequence_flags = flags();      if ((sequence_flags & DISABLE_ALL_PHASES) == DISABLE_ALL_PHASES) {        // If all the ports are disabled we should just fire the allocation        // done event and return.        done_signal_needed = true;        break;      }      if (!config || config->relays.empty()) {        // No relay ports specified in this config.        sequence_flags |= PORTALLOCATOR_DISABLE_RELAY;      }      if (!(sequence_flags & PORTALLOCATOR_ENABLE_IPV6) &&          networks[i]->GetBestIP().family() == AF_INET6) {        // Skip IPv6 networks unless the flag's been set.        continue;      }      // Disable phases that would only create ports equivalent to      // ones that we have already made.      DisableEquivalentPhases(networks[i], config, &sequence_flags);      if ((sequence_flags & DISABLE_ALL_PHASES) == DISABLE_ALL_PHASES) {        // New AllocationSequence would have nothing to do, so don't make it.        continue;      }      AllocationSequence* sequence =          new AllocationSequence(this, networks[i], config, sequence_flags);      if (!sequence->Init()) {//初始化udp_socket_:AsyncPacketSocket         delete sequence;        continue;      }      done_signal_needed = true;      //ports分配完毕后会触发BasicPortAllocatorSession::OnPortAllocationComplete 将事件抛给上层      sequence->SignalPortAllocationComplete.connect(          this, &BasicPortAllocatorSession::OnPortAllocationComplete);      if (running_)      //开始分配各种ports PHASE_UDP ->PHASE_RELAY->PHASE_TCP ->PHASE_SSLTCP        sequence->Start();      sequences_.push_back(sequence);    }  }  if (done_signal_needed) {    network_thread_->Post(this, MSG_SEQUENCEOBJECTS_CREATED);  }}//每种端口在分配时，会根据相应的协议获取Condidate将会通过SignalCandidatesReady信号通知到上层！//BasicPortAllocatorSession::SignalCandidatesReady==>//P2PTransportChannel::OnCandidatesReady==>P2PTransportChannel::SignalCandidateGathered==>//Transport::OnChannelCandidateGathered==>Transport::SignalCandidatesGathered==>//TransportController::OnTransportCandidatesGathered_w==>TransportController::SignalCandidatesGathered==>//WebRtcSession::OnTransportControllerCandidatesGathered==> ice_observer_->OnIceCandidate(&candidate);//最终会调用应用层实现的IceObserver.OnIceCandidate

在谷歌的WebRtc的demo中由PeerConnectionClient.java PCObserver实现，如下：

 private class PCObserver implements PeerConnection.Observer {    @Override    public void onIceCandidate(final IceCandidate candidate){      executor.execute(new Runnable() {        @Override        public void run() {          events.onIceCandidate(candidate);        }      });    }    @Override    public void onSignalingChange(        PeerConnection.SignalingState newState) {      Log.d(TAG, "SignalingState: " + newState);    }......}//events为PeerConnectionEvents 由CallActivity实现 public class CallActivity extends Activity    implements AppRTCClient.SignalingEvents,      PeerConnectionClient.PeerConnectionEvents,        CallFragment.OnCallEvents{        ......          @Override          public void onIceCandidate(final IceCandidate candidate) {            runOnUiThread(new Runnable() {              @Override              public void run() {                if (appRtcClient != null) {                  appRtcClient.sendLocalIceCandidate(candidate);                }              }            });          }        ......}public class WebSocketRTCClient implements AppRTCClient,    WebSocketChannelEvents{    .....      // Send Ice candidate to the other participant.          @Override          public void sendLocalIceCandidate(final IceCandidate candidate) {            executor.execute(new Runnable() {              @Override              public void run() {                JSONObject json = new JSONObject();                jsonPut(json, "type", "candidate");                jsonPut(json, "label", candidate.sdpMLineIndex);                jsonPut(json, "id", candidate.sdpMid);                jsonPut(json, "candidate", candidate.sdp);                if (initiator) {                  // Call initiator sends ice candidates to GAE server.                  if (roomState != ConnectionState.CONNECTED) {                    reportError("Sending ICE candidate in non connected state.");                    return;                  }                  //offer端通过http先将本地candidate发送到远程服务器，再由远程服务器发送到响应端！                  sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString());                  if (connectionParameters.loopback) {                    events.onRemoteIceCandidate(candidate);                  }                } else {                  // Call receiver sends ice candidates to websocket server.                  wsClient.send(json.toString());                }              }            });          }    .....    }

到目前位置ClientA端的工作告一段落，假如服务器服务器接受到响应端的SDP并转发给ClientA后，ClientA接下来会发生些什么呢？

SetRemoteSDP

服务器通过WebSocket将SDP发送到ClientA端后，最终会触发WebSocketChannelEvents.onWebSocketMessage方法，实现如下：

/*WebSocketRTCClient.java*/public class WebSocketRTCClient implements AppRTCClient,    WebSocketChannelEvents {    ......     @Override      public void onWebSocketMessage(final String msg) {        if (wsClient.getState() != WebSocketConnectionState.REGISTERED) {          Log.e(TAG, "Got WebSocket message in non registered state.");          return;        }        try {          JSONObject json = new JSONObject(msg);          String msgText = json.getString("msg");          String errorText = json.optString("error");          if (msgText.length() > 0) {            json = new JSONObject(msgText);            String type = json.optString("type");            //从服务器消息中获取到candidate            if (type.equals("candidate")) {              IceCandidate candidate = new IceCandidate(                  json.getString("id"),                  json.getInt("label"),                  json.getString("candidate"));              events.onRemoteIceCandidate(candidate);            } else if (type.equals("answer")) {              if (initiator) {              //Offer端收到了响应端的SDP                SessionDescription sdp = new SessionDescription(                    SessionDescription.Type.fromCanonicalForm(type),                    json.getString("sdp"));                events.onRemoteDescription(sdp);              } else {                reportError("Received answer for call initiator: " + msg);              }            } else if (type.equals("offer")) {              if (!initiator) {              //响应端收到了Offer端的SDP                SessionDescription sdp = new SessionDescription(                    SessionDescription.Type.fromCanonicalForm(type),                    json.getString("sdp"));                events.onRemoteDescription(sdp);              } else {                reportError("Received offer for call receiver: " + msg);              }            } else if (type.equals("bye")) {              events.onChannelClose();            } else {              reportError("Unexpected WebSocket message: " + msg);            }          } else {            if (errorText != null && errorText.length() > 0) {              reportError("WebSocket error message: " + errorText);            } else {              reportError("Unexpected WebSocket message: " + msg);            }          }        } catch (JSONException e) {          reportError("WebSocket message JSON parsing error: " + e.toString());        }      }    ......}

我们先看events.onRemoteDescription(sdp)的实现，后面再看 events.onRemoteIceCandidate(candidate)

events.onRemoteDescription(sdp)--> peerConnectionClient.setRemoteDescription(sdp);  public void setRemoteDescription(final SessionDescription sdp) {    executor.execute(new Runnable() {      @Override      public void run() {      ......        SessionDescription sdpRemote = new SessionDescription(            sdp.type, sdpDescription);        peerConnection.setRemoteDescription(sdpObserver, sdpRemote);     ......      }    });  }void PeerConnection::SetRemoteDescription(    SetSessionDescriptionObserver* observer,    SessionDescriptionInterface* desc) {  if (!VERIFY(observer != NULL)) {    LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";    return;  }  if (!desc) {    PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");    return;  }  // Update stats here so that we have the most recent stats for tracks and  // streams that might be removed by updating the session description.  stats_->UpdateStats(kStatsOutputLevelStandard);  std::string error;  //设置会话中的RemoteDescription，会根据RemoteDescription创建响应的channel  if (!session_->SetRemoteDescription(desc, &error)) {    PostSetSessionDescriptionFailure(observer, error);    return;  }  SetSessionDescriptionMsg* msg  = new SetSessionDescriptionMsg(observer);  //设置成功后调用相应的回调通知应用  signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);}bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,                                         std::string* err_desc) {  ASSERT(signaling_thread()->IsCurrent());  // Takes the ownership of |desc| regardless of the result.  rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);  // Validate SDP.  if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {    return false;  }  // Transport and Media channels will be created only when offer is set.  Action action = GetAction(desc->type());  //根据远程的SDP 创建会话需要的channel  if (action == kOffer && !CreateChannels(desc->description())) {    // TODO(mallinath) - Handle CreateChannel failure, as new local description    // is applied. Restore back to old description.    return BadRemoteSdp(desc->type(), kCreateChannelFailed, err_desc);  }  // Remove unused channels if MediaContentDescription is rejected.  RemoveUnusedChannels(desc->description());  // NOTE: Candidates allocation will be initiated only when SetLocalDescription  // is called.  //设置远程的description  set_remote_description(desc->description()->Copy());  if (!UpdateSessionState(action, cricket::CS_REMOTE, err_desc)) {    return false;  }......}

SetRemoteCondidate

 events.onRemoteIceCandidate(candidate)-->peerConnectionClient.addRemoteIceCandidate(candidate);--> peerConnection.addIceCandidate(candidate); bool PeerConnection::AddIceCandidate(    const IceCandidateInterface* ice_candidate) {  return session_->ProcessIceMessage(ice_candidate);}bool WebRtcSession::ProcessIceMessage(const IceCandidateInterface* candidate) {  if (state() == STATE_INIT) {     LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "                   << "without any offer (local or remote) "                   << "session description.";     return false;  }  if (!candidate) {    LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL";    return false;  }  bool valid = false;  if (!ReadyToUseRemoteCandidate(candidate, NULL, &valid)) {    if (valid) {      LOG(LS_INFO) << "ProcessIceMessage: Candidate saved";      saved_candidates_.push_back(          new JsepIceCandidate(candidate->sdp_mid(),                               candidate->sdp_mline_index(),                               candidate->candidate()));    }    return valid;  }  // Add this candidate to the remote session description.  if (!remote_desc_->AddCandidate(candidate)) {    LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used";    return false;  }  return UseCandidate(candidate);}UseCandidate--->transport_controller()->AddRemoteCandidates(content.name, candidates,                                                  &error)--->TransportController::AddRemoteCandidates_w--->Transport::AddRemoteCandidates--->void P2PTransportChannel::AddRemoteCandidate(const Candidate& candidate) {  ASSERT(worker_thread_ == rtc::Thread::Current());  uint32 generation = candidate.generation();  // Network may not guarantee the order of the candidate delivery. If a  // remote candidate with an older generation arrives, drop it.  if (generation != 0 && generation < remote_candidate_generation_) {    LOG(LS_WARNING) << "Dropping a remote candidate because its generation "                    << generation                    << " is lower than the current remote generation "                    << remote_candidate_generation_;    return;  }  // Create connections to this remote candidate.  //创建connections  CreateConnections(candidate, NULL);   // Resort the connections list, which may have new elements.  SortConnections();}

响应端的各个过程与此类似，只是相应的顺序不一样，结合Offer端不难明白！