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16 Commits

Author SHA1 Message Date
nisse
1bffc1d1a4 Rename rtc::Time64 --> rtc::TimeMillis. 
In the discussion on https://codereview.webrtc.org/1888593004/, a more
decriptive name was suggested for Time64.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/1923213002
Cr-Commit-Position: refs/heads/master@{#12594}
 2016-05-02 15:19:00 +00:00
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
e2af9ef638 Keep on sending stun binding requests on zero-cost networks. 
This is useful to keep the NAT binding alive on backup connections.

BUG=

Review URL: https://codereview.webrtc.org/1737493004

Cr-Commit-Position: refs/heads/master@{#11862}
 2016-03-03 16:27:53 +00:00
honghaiz
6b9ab9204b Cease all future TURN requests when a TURN refresh request fails for a given TURN port. 
This fixes an assert error in Turnport::OnSendStunPacket

BUG=webrtc:5388

Review URL: https://codereview.webrtc.org/1547373002

Cr-Commit-Position: refs/heads/master@{#11152}
 2016-01-05 17:06:20 +00:00
Honghai Zhang
8597543ae8 Schedule a CreatePermissionRequest after the success of a previous request 
unless a channel binding request is already scheduled.

BUG=5178
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1434603002 .

Cr-Commit-Position: refs/heads/master@{#10625}
 2015-11-12 19:07:25 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types. 
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
 2015-10-07 10:23:32 +00:00
Peter Thatcher
3e95d3ef39 Don't log warning for unexpected STUN binding responses. 
It was too spammy in the log because we have many code paths that check for responses when it's not a problem that it's not an expected response.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47199004

Cr-Commit-Position: refs/heads/master@{#9212}
 2015-05-18 22:55:06 +00:00
Peter Thatcher
1cf6f8101a Add logging for sending and receiving STUN binding requests and TURN requests and responses. 
BUG=
R=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46189004

Cr-Commit-Position: refs/heads/master@{#9195}
 2015-05-15 17:40:34 +00:00
andresp@webrtc.org
ff689be3c0 Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35079004

Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 11:55:32 +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
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