Commit Graph

13 Commits

Author SHA1 Message Date
1dca9d513a Support a user-provided string for the TLS ALPN extension.
Fix source formatting
Add TLS ALPN extension.

Bug: webrtc:8086
Change-Id: I1f28ccd78760d3415e465f734744d2c2f93845e2
Reviewed-on: https://chromium-review.googlesource.com/611150
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Diogo Real <diogor@google.com>
Cr-Commit-Position: refs/heads/master@{#19588}
2017-08-29 20:11:16 +00:00
786de70a59 Add TLS TURN tests.
This change extends the TurnPort tests to cover connections to
TURN servers over TLS.
As part of this, the TestTurnServer is extended to support
connections from clients over TLS.

Note that this also fixes the remaining bugs in webrtc:7562

Bug: webrtc:7584
Change-Id: If89ceae49d33417625464b5892d20eee4de7c3b5
Reviewed-on: https://chromium-review.googlesource.com/611520
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19397}
2017-08-17 23:03:04 +00:00
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
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
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
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
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
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
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
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
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
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