admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
11,877 Commits 1 Branch 0 Tags
Commit Graph

9 Commits

Author SHA1 Message Date
kwiberg
3ec4679dd2 Replace scoped_ptr with unique_ptr in webrtc/p2p/ 
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1923163003

Cr-Commit-Position: refs/heads/master@{#12532}
 2016-04-27 14:22:58 +00:00
jbauch
f1f87203d7 Split ByteBuffer into writer/reader objects. 
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.

BUG=webrtc:5155,webrtc:5670

Review URL: https://codereview.webrtc.org/1821083002

Cr-Commit-Position: refs/heads/master@{#12160}
 2016-03-30 13:43:44 +00:00
honghaiz
34b11eb66e Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p. 
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.

The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.

BUG=webrtc:5636

Review URL: https://codereview.webrtc.org/1793553002

Cr-Commit-Position: refs/heads/master@{#12019}
 2016-03-16 15:55:48 +00:00
honghaiz
c463e20069 Reset TURN port NONCE when a new socket is created. 
For example, when the TURN port has an ALLOCATE_MISMATCH error.

BUG=webrtc:5432

Review URL: https://codereview.webrtc.org/1595613004

Cr-Commit-Position: refs/heads/master@{#11453}
 2016-02-01 23:19:24 +00:00
deadbeef
376e1235c7 Destroy a Connection if a CreatePermission request fails. 
This means that if a TURN server denies permission for an
unreachable address, we'll no longer ping it fruitlessly.

BUG=webrtc:4917

Review URL: https://codereview.webrtc.org/1415313004

Cr-Commit-Position: refs/heads/master@{#10789}
 2015-11-25 17:00:12 +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