8cf9bdb3fa
Remove USE_WEBRTC_DEV_BRANCH.
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talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.
R=bjornv@webrtc.org , juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/39849004
Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:17:12 +00:00
dacdd9403d
Reland r7980:
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Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
BUG=4068, crbug/446908
R=juberti@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 17:33:34 +00:00
7e5b380437
Fix a crash in AllocationSequence.
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Internal bug 19074679.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8130 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 21:28:39 +00:00
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
9657265f39
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
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This reverts r7980.
It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.
Review URL: https://webrtc-codereview.appspot.com/41429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:08:27 +00:00
c5fd66dcdf
Accept incoming pings before remote answer is set to reduce connection latency.
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BUG=4068
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 19:23:37 +00:00
5ad4178137
Move the Jingle-specific network code into webrtc/libjingle.
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R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-23 22:14:15 +00:00
5647877b2d
Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
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R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 03:32:59 +00:00
18a3896bd2
Revert r7886:7887.
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Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org ,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
dee76f3b89
Move the obvious/easy Jingle-specific code into webrtc/libjingle.
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R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
33045ab2c1
Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003".
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BUG=3379
R=tpsiaki@google.com
Review URL: https://webrtc-codereview.appspot.com/27119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7674 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 19:43:11 +00:00
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
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
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
