Adding the ability to change ICE servers through SetConfiguration.

talk/app/webrtc/peerconnection.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 59 matching lines @@
 
 // PeerConnection implements the PeerConnectionInterface interface.
 // It uses WebRtcSession to implement the PeerConnection functionality.
 class PeerConnection : public PeerConnectionInterface,
                        public IceObserver,
                        public rtc::MessageHandler,
                        public sigslot::has_slots<> {
  public:
   explicit PeerConnection(PeerConnectionFactory* factory);
 
+  // TODO(deadbeef): Remove this overload of Initialize once everyone is moved
+  // to the new version.
   bool Initialize(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       const MediaConstraintsInterface* constraints,
       PortAllocatorFactoryInterface* allocator_factory,
       rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer);
+
+  bool Initialize(
+      const PeerConnectionInterface::RTCConfiguration& configuration,
+      const MediaConstraintsInterface* constraints,
+      rtc::scoped_ptr<cricket::PortAllocator> allocator,
+      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      PeerConnectionObserver* observer);
+
   rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
   rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
   bool AddStream(MediaStreamInterface* local_stream) override;
   void RemoveStream(MediaStreamInterface* local_stream) override;
 
   virtual WebRtcSession* session() { return session_.get(); }
 
   rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) override;
 
@@ skipping 293 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_