Merge remote bitrate estimator changes.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7811 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support for parsing header only RTP dumps with bwe_rtp_play.

Also adds support for printing the original_length in rtp_to_text.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d
Support new WebSocket signaling format.

- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.

BUG=3937,3995,4041
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.

BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d