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11 Commits

Author SHA1 Message Date
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base 
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
 2017-07-06 19:11:40 +00:00
Henrik Kjellander
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" 
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
 2017-07-01 14:48:18 +00:00
kjellander
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3) 
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
 2017-06-30 20:42:44 +00:00
Taylor Brandstetter
e5835f5d84 Adding an end-to-end connection time test. 
The test uses a fake clock and simulates network and signaling delays in
order to get a repeatable measurement of the time to establish a
connection (including DTLS). This will help ensure that various
optimizations continue to work as expected, and no new delays are
introduced.

This CL depends on: https://codereview.webrtc.org/2140283002/

R=honghaiz@webrtc.org, pthatcher@webrtc.org, skvlad@webrtc.org

Review URL: https://codereview.webrtc.org/2141863003 .

Cr-Commit-Position: refs/heads/master@{#14270}
 2016-09-16 22:07:58 +00:00
deadbeef
9794366ff0 Fixing memory leak in TurnServer. 
If the test TURN server received two allocate requests from the same
address, it was replacing the old allocation but not deleting it.

Also switching to std::unique_ptr to make it less likely for this to
pop up again.

Review-Url: https://codereview.webrtc.org/2114063002
Cr-Commit-Position: refs/heads/master@{#13449}
 2016-07-12 18:04:57 +00:00
Taylor Brandstetter
ef184702f6 Allow receiving a packet on a TURN port from an unknown address. 
This may occur if the TURN server allows the packet through due
to its policies around CreatePermission requirements, but the
candidate has not yet been signaled.

R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2086203004 .

Cr-Commit-Position: refs/heads/master@{#13278}
 2016-06-24 00:35:55 +00:00
Honghai Zhang
80f1db971d Include relay protocol type when computing the turn candidate foundation. 
BUG=576353
R=deadbeef@webrtc.org, pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1619213003 .

Cr-Commit-Position: refs/heads/master@{#11400}
 2016-01-27 19:54:44 +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
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