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Author SHA1 Message Date
Erik Språng
efdce6927e Disable some PortAllocatorTest on asan due to flakiness 
TBR=kjellander@webrtc.org
BUG=4743

Review URL: https://codereview.webrtc.org/1151173009

Cr-Commit-Position: refs/heads/master@{#9377}
 2015-06-05 07:41:36 +00:00
Erik Språng
85cf3c0794 Revert "Disable some PortAllocatorTest on tsan due to flakiness" 
This reverts commit 491bd534ef726456883ec372562d1c8fef82e7ca.

TBR=kjellander@webrtc.org
BUG=4743

Review URL: https://codereview.webrtc.org/1157743008

Cr-Commit-Position: refs/heads/master@{#9375}
 2015-06-05 07:31:13 +00:00
Erik Språng
491bd534ef Disable some PortAllocatorTest on tsan due to flakiness 
TBR=kjellander@webrtc.org
BUG=4743

Review URL: https://codereview.webrtc.org/1160033005

Cr-Commit-Position: refs/heads/master@{#9369}
 2015-06-04 12:57:59 +00:00
Jiayang Liu
d7e5c44e94 STUN allocation should not be disabled when using shared port and TURN servers are provided. 
BUG=
R=juberti@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48229004

Cr-Commit-Position: refs/heads/master@{#9091}
 2015-04-27 18:46:58 +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
pthatcher@webrtc.org
b4aac13810 Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49399004

Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 18:25:54 +00:00
guoweis@webrtc.org
931e0cf4b1 Fix WebRTC IP leaks. 
WebRTC binds to individual NICs and listens for incoming Stun packets. Sending stun through this specific NIC binding could make OS route the packet differently hence exposing non-VPN public IP.

The fix here is
1. to bind to any address (0:0:0:0) instead. This way, the routing will be the same as how chrome/http is.
2. also, remove the any all 0s addresses which happens when we bind to all 0s.

BUG=4276
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8418

Review URL: https://webrtc-codereview.appspot.com/39129004

Cr-Commit-Position: refs/heads/master@{#8419}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 19:10:22 +00:00
guoweis@webrtc.org
f358aea7bf Fix WebRTC IP leaks. 
WebRTC binds to individual NICs and listens for incoming Stun packets. Sending stun through this specific NIC binding could make OS route the packet differently hence exposing non-VPN public IP.

The fix here is
1. to bind to any address (0:0:0:0) instead. This way, the routing will be the same as how chrome/http is.
2. also, remove the any all 0s addresses which happens when we bind to all 0s.

BUG=4276
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39129004

Cr-Commit-Position: refs/heads/master@{#8418}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8418 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 18:44:14 +00:00
pbos@webrtc.org
8cf9bdb3fa Remove USE_WEBRTC_DEV_BRANCH. 
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
jiayl@webrtc.org
dacdd9403d Reland r7980: 
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
jiayl@webrtc.org
7e5b380437 Fix a crash in AllocationSequence. 
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
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
pthatcher@webrtc.org
9657265f39 Revert "Accept incoming pings before remote answer is set to reduce connection latency." 
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
jiayl@webrtc.org
c5fd66dcdf Accept incoming pings before remote answer is set to reduce connection latency. 
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
pthatcher@webrtc.org
5ad4178137 Move the Jingle-specific network code into webrtc/libjingle. 
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
pthatcher@webrtc.org
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. 
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
pbos@webrtc.org
18a3896bd2 Revert r7886:7887. 
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
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle. 
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
henrike@webrtc.org
33045ab2c1 Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003". 
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
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