59 Commits

Author SHA1 Message Date
deadbeef
c5d0d95fd8 Ensuring that UDP TURN servers are always used as STUN servers.
This was already working in most cases, but not for some corner cases:
* If the PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is not set
* If both a STUN server and TURN server are configured

I added unit tests for these cases, and centralized the code that gets
STUN server addresses in order to fix these and any related issues.

BUG=webrtc:4215

Review URL: https://codereview.webrtc.org/1215713003

Cr-Commit-Position: refs/heads/master@{#9596}
2015-07-16 17:22:28 +00:00
Peter Thatcher
b8b0143a11 Tighten link-local routing exclusion check
Also add a unit test for this behavior.

BUG=https://code.google.com/p/webrtc/issues/detail?id=4823
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1218293016 .

Cr-Commit-Position: refs/heads/master@{#9550}
2015-07-07 23:46:01 +00:00
Peter Thatcher
73ba7a690f Remove PORTALLOCATOR_ENABLE_BUNDLE, PortAllocatorSessionProxy, PortAllocatorSessionMuxer, and PortProxy.
R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46809004

Cr-Commit-Position: refs/heads/master@{#8999}
2015-04-14 16:25:58 +00:00
Guo-wei Shieh
be508a1d36 Implement Tcp Reconnect for TCPPort.
UDP case should not be changed.

Active TCPConnection will initiate Reconnect after OnClose and when Send or Ping fails.
Passive TCPConnection will prune itself as usual as the active side will create a new connection.

The Reconnect could make P2PCT choose a different best_connection in the case where connectivities exist b/w more than 1 Network.

Also, to avoid upper layer triggers ice restart, the WRITE_TIMEOUT caused by the socket disconnection is delayed  to give the reconnect mechanism chance to kick in. The timeout event is only fired if the reconnect can't work in 5 sec. If the reconnect, there should be no ICE disconnected state trigger either in active or passive side.

BUG=1926
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31359004

Cr-Commit-Position: refs/heads/master@{#8929}
2015-04-06 19:48:53 +00:00
pthatcher@webrtc.org
0ba1533fdb Added support for an Origin header in STUN messages.
For WebRTC there are instances where it may be desirable to provide
information to the STUN/TURN server about the website that initiated
a peer connection. This modification allows an origin string to be
included in the MediaConstraints object provided by the browser, which
is then passed as a STUN header in communications with the server.
A separate change will be submitted to the Chromium project that
uses and is dependent on this change, implementing IETF draft
http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02

Originally a patch from skobalt@gmail.com.

(https://webrtc-codereview.appspot.com/12839005/edit)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 00:47:02 +00:00
pkasting@chromium.org
332331fb01 Use uint16s for port numbers in webrtc/p2p/base.
This is a necessary precursor to using uint16s for port numbers more
consistently in Chromium code.

This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override).

BUG=chromium:81439
TEST=none
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 20:19:22 +00:00
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00