Add send transports to individual webrtc::Call streams.

webrtc/call.h

Index: webrtc/call.h
diff --git a/webrtc/call.h b/webrtc/call.h
index b94029db4730c143c22a9b84c6372af1ca9e6d0f..7bc9048485a65f518fa30d2565f31b51eccc87eb 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -69,16 +69,10 @@ class LoadObserver {
 class Call {
  public:
   struct Config {
-    Config() = delete;
-    explicit Config(newapi::Transport* send_transport)
-        : send_transport(send_transport) {}
+    Config() = default;

[pbos-webrtc, 2015/08/26 12:21:22]

Do you still want this line?

[the sun, 2015/08/27 08:42:21]

Not really.

 
     static const int kDefaultStartBitrateBps;
 
-    // TODO(solenberg): Need to add media type to the interface for outgoing
-    // packets too.
-    newapi::Transport* send_transport = nullptr;
-
     // VoiceEngine used for audio/video synchronization for this Call.
     VoiceEngine* voice_engine = nullptr;