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..97226a0e5ccba289159c17e4e03b3e72d83f5ce5 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -69,16 +69,8 @@ class LoadObserver {
 class Call {
  public:
   struct Config {
-    Config() = delete;
-    explicit Config(newapi::Transport* send_transport)
-        : send_transport(send_transport) {}
-
     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;