 269fb4bc90
			
		
	
	269fb4bc90
	
	
	
		
			
			Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder. BUG=3379 Review URL: https://webrtc-codereview.appspot.com/26999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
		
			
				
	
	
		
			176 lines
		
	
	
		
			6.9 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			176 lines
		
	
	
		
			6.9 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| /*
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|  * libjingle
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|  * Copyright 2004 Google Inc.
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|  *
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|  * Redistribution and use in source and binary forms, with or without
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|  * modification, are permitted provided that the following conditions are met:
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|  *
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|  *  1. Redistributions of source code must retain the above copyright notice,
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|  *     this list of conditions and the following disclaimer.
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|  *  2. Redistributions in binary form must reproduce the above copyright notice,
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|  *     this list of conditions and the following disclaimer in the documentation
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|  *     and/or other materials provided with the distribution.
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|  *  3. The name of the author may not be used to endorse or promote products
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|  *     derived from this software without specific prior written permission.
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|  *
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|  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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|  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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|  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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|  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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|  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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|  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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|  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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|  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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|  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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|  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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|  */
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| 
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| #ifndef TALK_SESSION_MEDIA_MEDIASESSIONCLIENT_H_
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| #define TALK_SESSION_MEDIA_MEDIASESSIONCLIENT_H_
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| 
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| #include <algorithm>
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| #include <map>
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| #include <string>
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| #include <vector>
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| #include "talk/media/base/cryptoparams.h"
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| #include "webrtc/p2p/base/session.h"
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| #include "webrtc/p2p/base/sessionclient.h"
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| #include "webrtc/p2p/base/sessiondescription.h"
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| #include "webrtc/p2p/base/sessionmanager.h"
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| #include "talk/session/media/call.h"
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| #include "talk/session/media/channelmanager.h"
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| #include "talk/session/media/mediasession.h"
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| #include "webrtc/base/messagequeue.h"
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| #include "webrtc/base/sigslot.h"
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| #include "webrtc/base/sigslotrepeater.h"
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| #include "webrtc/base/thread.h"
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| 
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| namespace cricket {
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| 
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| class Call;
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| 
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| class MediaSessionClient : public SessionClient, public sigslot::has_slots<> {
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|  public:
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| #if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
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|   MediaSessionClient(const buzz::Jid& jid, SessionManager *manager);
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| #endif
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|   // Alternative constructor, allowing injection of media_engine
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|   // and device_manager.
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|   MediaSessionClient(const buzz::Jid& jid, SessionManager *manager,
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|                      MediaEngineInterface* media_engine,
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|                      DataEngineInterface* data_media_engine,
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|                      DeviceManagerInterface* device_manager);
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|   ~MediaSessionClient();
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| 
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|   const buzz::Jid &jid() const { return jid_; }
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|   SessionManager* session_manager() const { return session_manager_; }
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|   ChannelManager* channel_manager() const { return channel_manager_; }
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| 
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|   // Return mapping of call ids to Calls.
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|   const std::map<uint32, Call *>& calls() const { return calls_; }
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| 
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|   // The settings below combine with the settings on SessionManager to choose
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| 
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|   // whether SDES-SRTP, DTLS-SRTP, or no security should be used. The possible
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|   // combinations are shown in the following table. Note that where either DTLS
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|   // or SDES is possible, DTLS is preferred. Thus to require either SDES or
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|   // DTLS, but not mandate DTLS, set SDES to require and DTLS to enable.
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|   //
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|   //              | SDES:Disable   | SDES:Enable    | SDES:Require   |
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|   // ----------------------------------------------------------------|
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|   // DTLS:Disable | No SRTP        | SDES Optional  | SDES Mandatory |
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|   // DTLS:Enable  | DTLS Optional  | DTLS/SDES Opt  | DTLS/SDES Mand |
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|   // DTLS:Require | DTLS Mandatory | DTLS Mandatory | DTLS Mandatory |
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| 
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|   // Control use of SDES-SRTP.
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|   SecurePolicy secure() const { return desc_factory_.secure(); }
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|   void set_secure(SecurePolicy s) { desc_factory_.set_secure(s); }
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| 
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|   // Control use of multiple sessions in a call.
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|   void set_multisession_enabled(bool multisession_enabled) {
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|     multisession_enabled_ = multisession_enabled;
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|   }
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| 
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|   int GetCapabilities() { return channel_manager_->GetCapabilities(); }
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| 
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|   Call *CreateCall();
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|   void DestroyCall(Call *call);
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| 
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|   Call *GetFocus();
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|   void SetFocus(Call *call);
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| 
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|   void JoinCalls(Call *call_to_join, Call *call);
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| 
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|   bool GetAudioInputDevices(std::vector<std::string>* names) {
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|     return channel_manager_->GetAudioInputDevices(names);
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|   }
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|   bool GetAudioOutputDevices(std::vector<std::string>* names) {
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|     return channel_manager_->GetAudioOutputDevices(names);
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|   }
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|   bool GetVideoCaptureDevices(std::vector<std::string>* names) {
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|     return channel_manager_->GetVideoCaptureDevices(names);
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|   }
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| 
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|   bool SetAudioOptions(const std::string& in_name, const std::string& out_name,
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|                        const AudioOptions& options) {
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|     return channel_manager_->SetAudioOptions(in_name, out_name, options);
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|   }
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|   bool SetOutputVolume(int level) {
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|     return channel_manager_->SetOutputVolume(level);
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|   }
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|   bool SetCaptureDevice(const std::string& cam_device) {
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|     return channel_manager_->SetCaptureDevice(cam_device);
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|   }
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| 
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|   SessionDescription* CreateOffer(const CallOptions& options) {
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|     return desc_factory_.CreateOffer(options, NULL);
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|   }
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|   SessionDescription* CreateAnswer(const SessionDescription* offer,
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|                                    const CallOptions& options) {
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|     return desc_factory_.CreateAnswer(offer, options, NULL);
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|   }
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| 
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|   sigslot::signal2<Call *, Call *> SignalFocus;
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|   sigslot::signal1<Call *> SignalCallCreate;
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|   sigslot::signal1<Call *> SignalCallDestroy;
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|   sigslot::repeater0<> SignalDevicesChange;
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| 
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|   virtual bool ParseContent(SignalingProtocol protocol,
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|                             const buzz::XmlElement* elem,
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|                             ContentDescription** content,
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|                             ParseError* error);
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|   virtual bool IsWritable(SignalingProtocol protocol,
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|                           const ContentDescription* content);
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|   virtual bool WriteContent(SignalingProtocol protocol,
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|                             const ContentDescription* content,
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|                             buzz::XmlElement** elem,
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|                             WriteError* error);
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| 
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|  private:
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|   void Construct();
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|   void OnSessionCreate(Session *session, bool received_initiate);
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|   void OnSessionState(BaseSession *session, BaseSession::State state);
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|   void OnSessionDestroy(Session *session);
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|   Session *CreateSession(Call *call);
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|   Session *CreateSession(const std::string& id, Call* call);
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|   Call *FindCallByRemoteName(const std::string &remote_name);
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| 
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|   buzz::Jid jid_;
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|   SessionManager* session_manager_;
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|   Call *focus_call_;
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|   ChannelManager *channel_manager_;
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|   MediaSessionDescriptionFactory desc_factory_;
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|   bool multisession_enabled_;
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|   std::map<uint32, Call *> calls_;
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| 
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|   // Maintain a mapping of session id to call.
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|   typedef std::map<std::string, Call *> SessionMap;
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|   SessionMap session_map_;
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| 
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|   friend class Call;
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| };
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| 
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| }  // namespace cricket
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| 
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| #endif  // TALK_SESSION_MEDIA_MEDIASESSIONCLIENT_H_
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