Enable SCTP compile for iOS.

Chromium's been updated to pull a version of usrsctplib that will compile correctly. This update DEPS to point at new revision and turn on the compile time flags for iOS sctp.

BUG=3211
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6681 4adac7df-926f-26a2-2b94-8c16560cd09d
Add a facility to the Thread class to catch blocking regressions.

This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.

This is a reland of an already reviewed cl (r6679) that got reverted by mistake.

TBR=xians@google.com,tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6682 4adac7df-926f-26a2-2b94-8c16560cd09d
Replace strcpy with talk_base::strcpyn.

Cpplint reports error 'Almost always, snprintf is better than strcpy'
when checking code styles. The function talk_base::strcpyn() is a better
option than strcpy().

BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12919004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6686 4adac7df-926f-26a2-2b94-8c16560cd09d
Handle the case if an unusually long peer name is provided in the peerconnection example.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove Thread::RunningForChannelManager().

I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
Minor refactoring of StatsCollector.
* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.

The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement unittest for SetSendCodecsChangesExistingStreams.

BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19869004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6699 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 71240799-> 71250251

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support of multiple STUN servers in UDPPort.
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.

I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.

BUG=3310
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
  m=  (media name and transport address)
  i=* (media title)
  c=* (connection information -- optional if included at
       session level)
  b=* (zero or more bandwidth information lines)
  k=* (encryption key)
  a=* (zero or more media attribute lines)

BUG=2260
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 6707 "Add support of multiple STUN servers in UDPPort."

Reason:
Breaks the build on callclient.cc.

> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
> 
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
> 
> BUG=3310
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13879004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
Move additional state into WebRtcVideoSendStream.

Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
fix

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.

This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.

BUG=3223
R=jiayl@webrtc.org, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d
Connect to the turn server if address cannot be resolved by the browser by using
unresolved address. This case is only considered for TCP sockets. P2P layer will
assume socket will do the resolve by using a proxy.

BUG=3384
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6722 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 71378257-> 71410012

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6726 4adac7df-926f-26a2-2b94-8c16560cd09d
Restart VideoReceiveStreams in WebRtcVideoEngine2.

Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.

Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.

BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d
Initial WebRtcVideoEngine2::GetStats().

Also forward-declaring and moving WebRtcVideoRenderer out of header.

BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
Creates the default track if the remote media content is send-only and there is no stream in the SDP.

BUG=2628
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 71452608-> 71453580

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d