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Author SHA1 Message Date
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
andrew@webrtc.org
4796cb93dc Disable flaky RelayServerTest.TestExpiration on all platforms. 
BUG=4134
TBR=pthatcher

Review URL: https://webrtc-codereview.appspot.com/37529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 23:56:19 +00:00
kjellander@webrtc.org
aeb0dd3079 Disable RelayServerTest.TestExpiration on Mac. 
The test is flaky on Mac.
BUG=4134
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-03 17:47:05 +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
pbos@webrtc.org
53cb74107f Make RelayServerTest use VirtualSocketServer. 
Permits running the tests in parallel.

R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w64 out/Debug/rtc_unittests --gtest_filter=RelayServerTest.*

Review URL: https://webrtc-codereview.appspot.com/38479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-22 07:56:42 +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
pthatcher@webrtc.org
aacc23465b Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
(This is the 3rd try)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 20:31:29 +00:00
guoweis@webrtc.org
4fba293c87 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
 2014-12-18 04:45:05 +00:00
pthatcher@webrtc.org
4cb3856a4d Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.

BUG=
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
This is an un-revert of r7992 and r7993.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 01:22:02 +00:00
guoweis@webrtc.org
c51fb9348d 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
 2014-12-18 00:30:55 +00:00
braveyao@webrtc.org
38881be912 If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport(). 
Verified in chromium. Now the existing content still could work.

BUG=4096
TEST=Manual Test
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7926 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 05:59:41 +00:00
guoweis@webrtc.org
950c518251 Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Committed: https://code.google.com/p/webrtc/source/detail?r=7906

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 23:01:31 +00:00
pthatcher@webrtc.org
f050791ba0 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7992.

It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c Split up (Jingle)Session from BaseSession. 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/35529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:37:37 +00:00
guoweis@webrtc.org
55360ae402 Revert "Add adapter_type into Candidate object." 
This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.

BUG=
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 05:28:10 +00:00
guoweis@webrtc.org
aaf02cc2d4 Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 23:03:10 +00:00
guoweis@webrtc.org
1f05c45976 Reenable test case P2PTransportChannelTest.TestIPv6Connections 
BUG=3317
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 21:25:54 +00:00
pbos@webrtc.org
fb108b5a28 Revert r7885. 
Breaks compile step of other code where network name of
cricket::Candidate is used.

TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 08:04:50 +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
e9db7fe80c Put pseudotcp back because remoting uses it. 
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-13 01:56:39 +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
guoweis@webrtc.org
8c9d79a29d Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 19:21:14 +00:00
guoweis@webrtc.org
8c9ff203c5 Redo the change of https://webrtc-codereview.appspot.com/30949004/ 
The previous change causes a build issue as there is subclass of TransportChannel in chromium. To break the circular dependency, a stub of implementation for GetState() is provided and will be removed once the jingle_glue::MockTransportChannel has the function defined.

TBR=pthatcher@webrtc.org

BUG=411086

Review URL: https://webrtc-codereview.appspot.com/34369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 07:56:02 +00:00
guoweis@webrtc.org
fd8422938c Revert "Implement GetState() for channel's connectivity check state." 
This reverts commit ff72f9e692d0918b32646dadaf382aa4355d8437.

TBR=pthatcher@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/33469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 00:51:59 +00:00
guoweis@webrtc.org
ff72f9e692 Implement GetState() for channel's connectivity check state. 
Previously, IceState is considered completed when there is only one connection (and the rest was trimmed). However, since the trimming logic is only done within the scope of network, when IPv6 and IPv4 both exist, the completion event is never fired.

This change adds the GetState() to each channel and it could decide what Completion means. The transport object then aggregates all channels before determining it's completed.

Each channel's IceState will be aggregrated at Transport level for overall Ice state

BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 00:14:49 +00:00
jiayl@webrtc.org
511f8a8ef2 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
 2014-12-03 02:17:07 +00:00
jiayl@webrtc.org
7806d8fe40 Fix an ASSERT that fires in a browser test for renegotiation. 
See https://code.google.com/p/chromium/issues/detail?id=293125#c33

BUG=crbug/293125
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 19:58:50 +00:00
guoweis@webrtc.org
930e004a81 Add jmi field for packets discarded due to network error 
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7693

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 19:42:14 +00:00
henrike@webrtc.org
6a782c2a46 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. 
TBR=guoweis@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 22:33:13 +00:00
guoweis@webrtc.org
312614a438 Add jmi field for packets discarded due to network error 
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 03:38:05 +00:00
henrike@webrtc.org
43e033e778 Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted." 
BUG=3379
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 19:40:29 +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