| d59daf8023286d63a1b6c8af82eedb684181c1eb |
|
15-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Merging BaseSession code into WebRtcSession.
After the TransportController CL, BaseSession does little more than
hold a state and an error, and act as an intermediary for the
TransportController. So it doesn't make sense for it to be its own
class.
Review URL: https://codereview.webrtc.org/1397973002
Cr-Commit-Position: refs/heads/master@{#10281}
/external/webrtc/webrtc/p2p/base/session.h
|
| 1f429e34180ca19a7fb98b89bacd34d42e9b01ec |
|
28-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Passing the new policy from PeerConnection RTCConfiguration to
p2ptransportchannel. This CL does not use the new policy yet.
BUG=
Review URL: https://codereview.webrtc.org/1369773003
Cr-Commit-Position: refs/heads/master@{#10092}
/external/webrtc/webrtc/p2p/base/session.h
|
| cbecd358e032021eac11fb13e04ec7f070d4f407 |
|
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/
Which fixes the FYI bots for the original CL, and breaks them for this revert.
Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}
TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1361773005
Cr-Commit-Position: refs/heads/master@{#10036}
/external/webrtc/webrtc/p2p/base/session.h
|
| a81a42f584baa0d93a4b93da9632415e8922450c |
|
23-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1358413003
Cr-Commit-Position: refs/heads/master@{#10024}
/external/webrtc/webrtc/p2p/base/session.h
|
| 47ee2f3b9f33e8938948c482c921d4e13a3acd83 |
|
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
Review URL: https://codereview.webrtc.org/1350523003
Cr-Commit-Position: refs/heads/master@{#10022}
/external/webrtc/webrtc/p2p/base/session.h
|
| 8902433a43bbc9cc0de4966774d3dbbe37ef96fb |
|
18-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.
Cr-Commit-Position: refs/heads/master@{#9994}
/external/webrtc/webrtc/p2p/base/session.h
|
| 9af63f473e1d0d6c47a741a046c41642dfc1c178 |
|
18-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".
Review URL: https://codereview.webrtc.org/1246913005
Cr-Commit-Position: refs/heads/master@{#9993}
/external/webrtc/webrtc/p2p/base/session.h
|
| 7cbd188c5ed7df80bb737bd4ada94422730e2d89 |
|
18-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (again).
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1353713002 .
Cr-Commit-Position: refs/heads/master@{#9979}
/external/webrtc/webrtc/p2p/base/session.h
|
| d12140a68efdcffa1c2c18f25149905e9dae1a9c |
|
10-Sep-2015 |
guoweis <guoweis@webrtc.org> |
Revert change which removes GICE.
There are still dependencies on this functionality.
TBR=pthatcher@webrtc.org
BUG=526399
Review URL: https://codereview.webrtc.org/1336553003
Cr-Commit-Position: refs/heads/master@{#9920}
/external/webrtc/webrtc/p2p/base/session.h
|
| d82819892a382899a82ced756a9922a84ca9ca98 |
|
27-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer.
Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.
This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1312643004 .
Cr-Commit-Position: refs/heads/master@{#9794}
/external/webrtc/webrtc/p2p/base/session.h
|
| 2159b89fa2cb55beeef38f72bd45e217f3d33d4e |
|
22-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
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 5bdafd44c86ee46bd7e040f19828324583418b33.
Original CL: https://codereview.webrtc.org/1263663002/
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1303393002 .
Cr-Commit-Position: refs/heads/master@{#9761}
/external/webrtc/webrtc/p2p/base/session.h
|
| 5bdafd44c86ee46bd7e040f19828324583418b33 |
|
21-Aug-2015 |
minyuel <minyue@webrtc.org> |
Revert "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 081f34b564e1a26ffbbe9515eba1fef7c736fdde.
Original code review see
https://codereview.webrtc.org/1291363005
The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137
TBR=pthatcher@webrtc.org,
BUG=
Review URL: https://codereview.webrtc.org/1308753003 .
Cr-Commit-Position: refs/heads/master@{#9756}
/external/webrtc/webrtc/p2p/base/session.h
|
| 081f34b564e1a26ffbbe9515eba1fef7c736fdde |
|
20-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
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}
/external/webrtc/webrtc/p2p/base/session.h
|
| fa301809b698017455847f45cc7e0dfa1bdfed35 |
|
11-Aug-2015 |
pthatcher <pthatcher@webrtc.org> |
Revert "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 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
/external/webrtc/webrtc/p2p/base/session.h
|
| 3449faa553ec94c52ef2d0949867befb60992c88 |
|
10-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
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}
/external/webrtc/webrtc/p2p/base/session.h
|
| 900996290c996193ac3e418f315354fd2bd0ea8a |
|
13-Jul-2015 |
honghaiz <honghaiz@webrtc.org> |
Add methods to set the ICE connection receiving_timeout values.
BUG=
Review URL: https://codereview.webrtc.org/1231913003
Cr-Commit-Position: refs/heads/master@{#9572}
/external/webrtc/webrtc/p2p/base/session.h
|
| a6d2444c84004d10a5d8b8517bbd178600f8412f |
|
10-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1228203002 .
Cr-Commit-Position: refs/heads/master@{#9564}
/external/webrtc/webrtc/p2p/base/session.h
|
| 54360510ff9b7c61fc906d3ed360b06a5824bbf1 |
|
08-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
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}
/external/webrtc/webrtc/p2p/base/session.h
|
| 04e5b498278c633bc3c49da43d08c15b1e75ebc0 |
|
29-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Make maximum SSL version configurable through PeerConnectionFactory::Options
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}
/external/webrtc/webrtc/p2p/base/session.h
|
| 0e209b03bf55d6daf209e35b3a8e8b6eab3d4d52 |
|
24-Mar-2015 |
Donald Curtis <decurtis@webrtc.org> |
Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/.
BUG=1574
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36659004
Cr-Commit-Position: refs/heads/master@{#8851}
/external/webrtc/webrtc/p2p/base/session.h
|
| 592470b4ff39d60b52c745432ec131f05f3b6aa9 |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47599004
Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| 6ad507ac35ce638beddd7ac6687d006995637253 |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE.
Also, remove channel_name. It's no longer needed.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43719004
Cr-Commit-Position: refs/heads/master@{#8741}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8741 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| c04a97f054348909c5b0c24369fb4272c2c16041 |
|
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move from BaseSession::GetStats to WebRtcSession::GetTransportStats
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
Review URL: https://webrtc-codereview.appspot.com/45639004
Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| 990a00c30a2e87972506aac3a992a93ed3c8f79a |
|
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove unused transport code.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49389004
Cr-Commit-Position: refs/heads/master@{#8719}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8719 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| 877ac765ad30a22148da41695fa607682af4a191 |
|
04-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38909004
Cr-Commit-Position: refs/heads/master@{#8245}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| 520a69e8ea71e93528f258b1c2f85d1660fe9647 |
|
04-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 8238 "Add RefCounting for TransportProxies"
Failing on Mac64_Debug
> Add RefCounting for TransportProxies
>
> BUG=1574
> R=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/37869004
TBR=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37159004
Cr-Commit-Position: refs/heads/master@{#8243}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8243 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| c5f697135e626044b15eacdc82fd840fbe74b351 |
|
04-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 8237 "Cleanup and prepare for bundling."
libjingle_peerconnection_objc_test consistently failing on Mac64 Debug.
> Cleanup and prepare for bundling.
>
> - Add a GetOptions function. Needed for eventual bundle testing to
> confirm that channel options are preserved.
> - Simplify unit tests and cleanup unused code.
>
> BUG=1574
> R=pthatcher@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/39699004
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34959004
Cr-Commit-Position: refs/heads/master@{#8241}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| e2506670a4e57cbd351141d8ccf7635ffd2db093 |
|
04-Feb-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Add RefCounting for TransportProxies
BUG=1574
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37869004
Cr-Commit-Position: refs/heads/master@{#8238}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8238 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| af01d93aa2d75b39cdcaadd682c5c60336c75ea7 |
|
04-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
BUG=1574
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39699004
Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| 357469da5a2d9d3f03d50a56e12a9a5166c3c948 |
|
15-Jan-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels.
Until the TransportProxy enters the "negotiated" state we only create
ChannelImpls but we don't hook up to them. However, we still neeed to
reserve their spot and increment the reference count.
Once we are negotiated we can hook all the ChannelProxy's to the
corresponding ChannelImpls.
This change is needed to implement maxbundle.
BUG=1574
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8082 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/session.h
|
| aacc23465b72151fece2e6836a7c43463d3ed41d |
|
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
|
| 4cb3856a4d4782cc7abf228a7f01ea70812d9fb1 |
|
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
|
| 536f999e58ee7456d116afad734aa64d548f1a49 |
|
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| f050791ba071eb208da4e95abc2ff21f57d0738f |
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16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| 4afb59903c2dcc893cd86a973cc16da4201e387c |
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16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| 18a3896bd28b63fa35168cd6c8d41c8cebaab3dd |
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15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| dee76f3b89b9339699e0321a3afc643ee06afa09 |
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12-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
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28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
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18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
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| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
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15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/session.h
|