admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
14,320 Commits 1 Branch 0 Tags
Commit Graph

471 Commits

Author SHA1 Message Date
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
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
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