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7f04b08d3b
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Issue 4780: disabling multiple_routes breaks Turn/Tcp.
BUG=webrtc:4780
R=pthatcher@chromium.org
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1196453005.
Cr-Commit-Position: refs/heads/master@{#9473}
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2015-06-19 18:27:16 +00:00 |
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1cf6f8101a
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Add logging for sending and receiving STUN binding requests and TURN requests and responses.
BUG=
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46189004
Cr-Commit-Position: refs/heads/master@{#9195}
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2015-05-15 17:40:34 +00:00 |
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b32a5c48d3
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Add more logging around TURN refreshes.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50669004
Cr-Commit-Position: refs/heads/master@{#8979}
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2015-04-10 21:04:45 +00:00 |
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fe672e3839
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release the turn allocation by sending a refresh request with lifetime 0
BUG=406578
Patch originally from philipp.hancke@googlemail.com
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8087 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-01-17 00:58:15 +00:00 |
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19e4e8d751
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Add support for trying alternate server (STUN 300 error message) on TCP
BUG=3774
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8036 4adac7df-926f-26a2-2b94-8c16560cd09d
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2015-01-10 02:41:32 +00:00 |
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0ba1533fdb
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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
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2015-01-10 00:47:02 +00:00 |
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4fba293c87
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Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
BUG=3927
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7941 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-12-18 04:45:05 +00:00 |
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c51fb9348d
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Fix an assert failure caused by race condition
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7938 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-12-18 00:30:55 +00:00 |
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511f8a8ef2
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TurnPort should ignore STUN binding reponses when using shared socket.
BUG=4043
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7792 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-12-03 02:17:07 +00:00 |
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332331fb01
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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
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2014-11-06 20:19:22 +00:00 |
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269fb4bc90
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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
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2014-10-28 22:20:11 +00:00 |
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28100cb388
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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
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2014-10-17 22:03:39 +00:00 |
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d1ba6d9cbf
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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
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2014-10-15 17:30:28 +00:00 |
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