63 Commits

Author SHA1 Message Date
pthatcher
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.
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
Peter Thatcher
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
honghaiz
b19eba3d4b Fix Turn TCP port issue.
Sometimes the port still try to send stun packet when the connection is disconnected,
causing an assertion error.

BUG=4859

Review URL: https://codereview.webrtc.org/1247573002

Cr-Commit-Position: refs/heads/master@{#9671}
2015-08-03 17:23:40 +00:00
kwiberg@webrtc.org
eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-24 09:20:19 +00:00
bjornv@webrtc.org
95a32ec098 Revert 8271 "VirtualSocketServer out-of-order issue with closing..."
Failed on Linux_Memcheck bot.
http://chromegw/i/client.webrtc/builders/Linux%20Memcheck/builds/3182

> VirtualSocketServer out-of-order issue with closing TCP sockets
> 
> https://webrtc-codereview.appspot.com/41449004 added a TURN TCP
> allocation release test which was disabled as it triggered an assert
> in the turnserver.
> 
> This was caused by VirtualSockerServer delivering the last TCP packet
> after closing the connection. Calling
>     VirtualSocketServer::SendTcp
> and
>     VirtualSocket::Close
> from TestTurnTCPReleaseAllocation led to the following order of
> messages in VirtualSocket::OnMessage:
>     MSG_ID_DISCONNECT
>     MSG_ID_PACKET
> 
> This is out of order and triggers an assert in turnserver.cc since the
> socket from which the message arrives has already been discarded,
> subsequently breaking the test.
> 
> In VirtualSocketServer::Disconnect the MSG_ID_DISCONNECT is posted to the
> msg_queue immediately, thus getting ahead of any (slightly delayed)
> actual packets.
> 
> Maybe PostAt(network_delay_ + 1, ...) would be better?
> 
> Re-enables TestTurnTCPReleaseAllocation.
> 
> BUG=
> R=juberti@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/34759004

TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38979004

Cr-Commit-Position: refs/heads/master@{#8280}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8280 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-07 06:47:21 +00:00
pthatcher@webrtc.org
4770437da9 VirtualSocketServer out-of-order issue with closing TCP sockets
https://webrtc-codereview.appspot.com/41449004 added a TURN TCP
allocation release test which was disabled as it triggered an assert
in the turnserver.

This was caused by VirtualSockerServer delivering the last TCP packet
after closing the connection. Calling
    VirtualSocketServer::SendTcp
and
    VirtualSocket::Close
from TestTurnTCPReleaseAllocation led to the following order of
messages in VirtualSocket::OnMessage:
    MSG_ID_DISCONNECT
    MSG_ID_PACKET

This is out of order and triggers an assert in turnserver.cc since the
socket from which the message arrives has already been discarded,
subsequently breaking the test.

In VirtualSocketServer::Disconnect the MSG_ID_DISCONNECT is posted to the
msg_queue immediately, thus getting ahead of any (slightly delayed)
actual packets.

Maybe PostAt(network_delay_ + 1, ...) would be better?

Re-enables TestTurnTCPReleaseAllocation.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34759004

Cr-Commit-Position: refs/heads/master@{#8271}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8271 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 16:33:47 +00:00
pthatcher@webrtc.org
fe672e3839 release the turn allocation by sending a refresh request with lifetime 0
BUG=406578

Patch originally from philipp.hancke@googlemail.com

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8087 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-17 00:58:15 +00:00
guoweis@webrtc.org
19e4e8d751 Add support for trying alternate server (STUN 300 error message) on TCP
BUG=3774
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8036 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 02:41:32 +00:00
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
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
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