365 Commits

Author SHA1 Message Date
jiayl@webrtc.org
e21cc9ae2a When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated.
constraints . SetMandatoryReceiveAudio (false);

The problem is that webrtc::GetTrackIdBySsrc returns false if audio is not available. However it should continue and check for the video track.

BUG=webrtc:3755
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7005 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 22:21:34 +00:00
niklas.enbom@webrtc.org
4431fd6ad5 Add 60 fps video support
R=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7000 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 14:57:46 +00:00
solenberg@webrtc.org
00f11f5e24 - Make local constant non-static.
- Remove spammy log line.

BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 08:52:17 +00:00
pbos@webrtc.org
b648b9d85c Remove test constructor in WebRtcVideoEngine2.
Removes the need for ::Construct().

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 11:08:06 +00:00
kjellander@webrtc.org
e9bfed0648 Move constant so it is not stripped out for TSAN bots.
BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 19:46:26 +00:00
solenberg@webrtc.org
6556a59db1 As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests.
Also, caused some issues with other peerconnection_unittest tests, so changed the design of those.

BUG=
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6968 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:35:40 +00:00
buildbot@webrtc.org
b4c7b09c13 (Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 12:11:58 +00:00
buildbot@webrtc.org
2b0554f0e7 (Auto)update libjingle 73794259-> 73891518
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-22 14:08:15 +00:00
henrike@webrtc.org
0481f15f02 (Auto)update libjingle 73399579-> 73626167
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 14:56:59 +00:00
tommi@webrtc.org
5b06b06cc0 Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."
The bot that had the problem was using an old version of STL, so relanding.

> Revert 6863 "Refactor StatsCollector and associated types."
> 
> Breaks chrome compilation on Mac:
> 
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
> error: no matching constructor for initialization of
> 'webrtc::StatsReport'
>           _Tp __x_copy = __x;
>               ^          ~~~
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::_M_insert_aux' requested here
>           _M_insert_aux(end(), __x);
>           ^
> ../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::push_back' requested here
>   reports.push_back(report1);
>           ^
> ../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
> note: candidate constructor not viable: requires 0 arguments, but 1
> was provided
>   StatsReport() : timestamp(0) {}
> 
> 
> 
> > Refactor StatsCollector and associated types.
> > * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> > * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> > * Report ids are now const.
> > * Copying of data has been greatly reduced.
> > * This change includes preparation work for making GetStats fully async.
> > 
> > This is a reland of r6778 which was reverted due to fyi bots failing.
> > I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> > 
> > R=xians@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/15119004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/21169004

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-15 08:38:30 +00:00
niklas.enbom@webrtc.org
22fa032f22 Revert 6863 "Refactor StatsCollector and associated types."
Breaks chrome compilation on Mac:

/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
          _Tp __x_copy = __x;
              ^          ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
          _M_insert_aux(end(), __x);
          ^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
  reports.push_back(report1);
          ^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
  StatsReport() : timestamp(0) {}



> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
> 
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15119004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 23:11:04 +00:00
henrike@webrtc.org
d968dd039a Fixes failure triggered by include order re-ordering.
BUG=N/A
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 18:39:43 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +00:00
xians@webrtc.org
38d88816e3 Fix the audio source failure due to unsupported constraints.
Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.

This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.

BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback

R=henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 13:51:58 +00:00
mallinath@webrtc.org
e999bd087b Removing ASSERT for tcp candidate for port 0 and 9, as Android clients
may not be called with set_allow_tcp_listen(false).

This CL will also sends tcp candidate in RFC 6544 format.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 06:05:55 +00:00
tommi@webrtc.org
730bf30da7 Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11 14:08:33 +00:00
jiayl@webrtc.org
7ec3f9f838 Fix a bug in parsing IceCandidate with IPV6 address.
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.

BUG=3669
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-08 23:09:15 +00:00
mallinath@webrtc.org
2d60c5e8bc Encoding and Decoding of TCP candidates as defined in RFC 6544.
R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204

Review URL: https://webrtc-codereview.appspot.com/21479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-08 22:29:20 +00:00
buildbot@webrtc.org
53df88c1bc (Auto)update libjingle 72847605-> 72850595
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 22:46:01 +00:00
jiayl@webrtc.org
e7d47a1473 Maintain the order of the m-lines in CreateOffer and CreateAnswer.
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.

BUG=2395
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-05 19:19:05 +00:00
jiayl@webrtc.org
b18bf5e47d Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer.
Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.

BUG=3282
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 18:34:16 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
buildbot@webrtc.org
51c5508bf1 (Auto)update libjingle 72016417-> 72097588
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-28 22:26:15 +00:00
henrike@webrtc.org
185636cf70 Revert of 6778 "Refactor StatsCollector and associated types."
Breakes FYI bots.

BUG=N/A
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6783 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 18:44:42 +00:00
tommi@webrtc.org
190d269c0f Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

(This is a reland of the original attempt in r6747)

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6778 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 10:32:30 +00:00
jiayl@webrtc.org
06b04ec4ab Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.
BUG=3579
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6776 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 20:41:20 +00:00
buildbot@webrtc.org
45304ff0a7 (Auto)update libjingle 71829282-> 71834788
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 16:06:35 +00:00
henrike@webrtc.org
39f831fbb0 Re-revert of 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.

BUG=N/A
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 14:20:52 +00:00
buildbot@webrtc.org
437d57db5b (Auto)update libjingle 71775619-> 71778545
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:40:28 +00:00
henrike@webrtc.org
8c7e3291a9 Revert 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.

BUG=N/A
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:38:58 +00:00
henrike@webrtc.org
8721f989bf Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."
BUG=N/A
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6769 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:38:09 +00:00
buildbot@webrtc.org
e2da234e27 (Auto)update libjingle 71766184-> 71775619
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:09:01 +00:00
tommi@webrtc.org
0f7328cd6b Temporarily add a default ctor to StatsReport and make |id| non const.
As soon as I've updated the chrome side, I'll revert this cl.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/16149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6766 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 16:31:57 +00:00
glaznev@webrtc.org
efe4b9af49 Add VP8 video decoding hw acceleration support to Java Peerconnection library.
For now NVidia decoder is supported only,
Qualcomm will be added once b/16353967 is fixed.

TODO:
- Support queuing 2-3 decoder input buffers.
- Add average decoding time statistics.
- Add Qualcomm hw decoder support.

BUG=3030
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6758 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 17:44:53 +00:00
buildbot@webrtc.org
e69b061926 (Auto)update libjingle 71575585-> 71599033
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 20:38:58 +00:00
tommi@webrtc.org
908f57ed94 Disable GetStatsForInvalidTrack while I rewrite it.
TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6748 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:44:39 +00:00
tommi@webrtc.org
756b8462eb Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

R=xians@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6745

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:24:17 +00:00
tommi@webrtc.org
fd61a1d693 Revert 6745 "Refactor StatsCollector and associated types."
Broke build on android.

> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18819004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:05:28 +00:00
tommi@webrtc.org
647e05cfcd Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 10:55:11 +00:00
jiayl@webrtc.org
3edbaaf337 Ignore empty data in DataChannel::Send to match FF's behavior.
BUG=crbug/395205
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6742 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 23:57:50 +00:00
jiayl@webrtc.org
a0b929b63c Revert "Reland r6707 with the fix for callclient.cc."
Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.

TBR=wu@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/17979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 22:28:36 +00:00
buildbot@webrtc.org
3dec81a736 (Auto)update libjingle 71456173-> 71456344
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:39:56 +00:00
jiayl@webrtc.org
a6e8cf8fb7 Reland r6707 with the fix for callclient.cc.
TBR=mallinath@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/13039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:34:11 +00:00
buildbot@webrtc.org
60e65b11c1 (Auto)update libjingle 71452608-> 71453580
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:07:50 +00:00
jiayl@webrtc.org
8636fc852e Creates the default track if the remote media content is send-only and there is no stream in the SDP.
BUG=2628
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 20:54:27 +00:00
mallinath@webrtc.org
e5995aadd5 Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.
This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.

BUG=3223
R=jiayl@webrtc.org, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 18:23:52 +00:00
jiayl@webrtc.org
e10d28cf14 fix
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 17:07:49 +00:00
wu@webrtc.org
52eddec71b Revert 6707 "Add support of multiple STUN servers in UDPPort."
Reason:
Breaks the build on callclient.cc.

> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
> 
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
> 
> BUG=3310
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13879004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 00:03:24 +00:00
wu@webrtc.org
4c3e9917e7 Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
m=  (media name and transport address)
  i=* (media title)
  c=* (connection information -- optional if included at
       session level)
  b=* (zero or more bandwidth information lines)
  k=* (encryption key)
  a=* (zero or more media attribute lines)

BUG=2260
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:03:13 +00:00
jiayl@webrtc.org
46fb331bc5 Add support of multiple STUN servers in UDPPort.
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.

I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.

BUG=3310
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 20:55:31 +00:00