Peter Thatcher
b8b0143a11
Tighten link-local routing exclusion check
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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.
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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.
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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.
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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.
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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
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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."
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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.
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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
