081f34b564
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
...
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1291363005 .
Cr-Commit-Position: refs/heads/master@{#9738}
2015-08-20 03:37:59 +00:00
fa301809b6
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
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This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org , juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
2015-08-11 11:13:00 +00:00
3449faa553
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
...
R=deadbeef@webrtc.org , juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1263663002 .
Cr-Commit-Position: refs/heads/master@{#9692}
2015-08-10 19:22:59 +00:00
900996290c
Add methods to set the ICE connection receiving_timeout values.
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BUG=
Review URL: https://codereview.webrtc.org/1231913003
Cr-Commit-Position: refs/heads/master@{#9572}
2015-07-13 19:19:42 +00:00
54360510ff
Add flakyness check based on the recently received packets.
...
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1207563002 .
Cr-Commit-Position: refs/heads/master@{#9553}
2015-07-08 18:08:39 +00:00
04e5b49827
Make maximum SSL version configurable through PeerConnectionFactory::Options
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This can be used to activate DTLS 1.2 through a command-line flag from Chromium
later.
BUG=chromium:428343
R=jiayl@webrtc.org , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/54509004
Cr-Commit-Position: refs/heads/master@{#9328}
2015-05-29 07:40:51 +00:00
ae0f0ee79e
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
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Just use the less-evil version, DISALLOW_COPY_AND_ASSIGN macro.
This should help with my TODO in
https://chromium.googlesource.com/chromium/src/+/master/base/macros.h#33
Tested on Linux with the following command lines:
$ rm -rf out/
$ gn gen //out/Debug --args='is_debug=true target_cpu="x64" build_with_chromium=false'
$ ninja -C out/Debug
BUG=None
TEST=see above
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50599004
Patch from Thiago Farina <tfarina@chromium.org >.
Cr-Commit-Position: refs/heads/master@{#8927}
2015-04-04 23:56:56 +00:00
245989b22a
Address comments from cr 43769004.
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- Remove unnecessary hop to worker from OnChannelRequestSignaling_s.
- Remove now-not-needed component param.
- Update documentation.
R=juberti@webrtc.org
BUG=4444
Review URL: https://webrtc-codereview.appspot.com/42839004
Cr-Commit-Position: refs/heads/master@{#8852}
2015-03-24 16:56:34 +00:00
462dbcfc2a
Fix bug in Transport where channel_.clear() was being called without a lock.
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Looks like this snuck in between misaligned braces.
Also switching to C++11 for loops, reducing lock scopes in a few places and removing locks in others.
BUG=4444
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43769004
Cr-Commit-Position: refs/heads/master@{#8765}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8765 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 21:40:26 +00:00
3ee4fe5a94
Re-land: Add API to get negotiated SSL ciphers
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This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
The previously approved CL https://webrtc-codereview.appspot.com/26009004/ was reverted in https://webrtc-codereview.appspot.com/40689004/ due to compilation issues while rolling into Chromium.
As the new method has landed in Chromium in https://crrev.com/bc321c76ace6e1d5a03440e554ccb207159802ec , this should be safe to land here now.
BUG=3976
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37209004
Cr-Commit-Position: refs/heads/master@{#8343}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8343 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 22:35:30 +00:00
2bf0e90c9d
Revert 8275 "This CL adds an API to the SSL stream adapters and ..."
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I'm reverting the patch due to compilation issues. It would be great if we could make sure Chromium is ready for the patch before we land it in WebRTC.
As is, we're trying to roll webrtc into Chromium and we can't (this is not the only reason though). I might reland this after the roll, depending on how that goes though.
Here's an example failure:
e:\b\build\slave\win_gn\build\src\jingle\glue\channel_socket_adapter_unittest.cc(77) : error C2259: 'jingle_glue::MockTransportChannel' : cannot instantiate abstract class
due to following members:
'bool cricket::TransportChannel::GetSslCipher(std::string *)' : is abstract
e:\b\build\slave\win_gn\build\src\third_party\webrtc\p2p\base\transportchannel.h(107) : see declaration of 'cricket::TransportChannel::GetSslCipher'
ninja: build stopped: subcommand failed.
> This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
>
> BUG=3976
> R=davidben@chromium.org , juberti@webrtc.org , pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26009004
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40689004
Cr-Commit-Position: refs/heads/master@{#8282}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8282 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-07 11:13:18 +00:00
1d11c8202b
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
...
BUG=3976
R=davidben@chromium.org , juberti@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26009004
Cr-Commit-Position: refs/heads/master@{#8275}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8275 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 19:47:39 +00:00
a907e01c63
Adding constness.
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Make a few member variables in the Transport class officially const so that it's clear that locking isn't needed for access. There are getters for some of these (e.g. content_name()) that don't have locking or checking, so making the variables const is at least a way to guard against regressions. Also making the clock_ member in overuse_frame_detector.h const for clarity that it doesn't require a lock for access.
No code change.
Review URL: https://webrtc-codereview.appspot.com/35949004
Cr-Commit-Position: refs/heads/master@{#8186}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8186 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 17:33:45 +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.
...
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
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
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."
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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
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.
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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
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."
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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
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
930e004a81
Add jmi field for packets discarded due to network error
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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
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
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
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."
...
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.
...
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
