Commit Graph

11 Commits

Author SHA1 Message Date
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
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
38f8893235 WebRTC Bug 4865
Bug 4865: even without STUN/TURN, as long as the peer is on the open internet, the connectivity should work. This is actually a regression even for hangouts.

We need to issue the 0.0.0.0 candidate into Port::candidates_ and filter it out later. The reason is that when we create connection, we need a local candidate to match the remote candidate.

The same connection later will be updated with the prflx local candidate once the STUN ping response is received.

BUG=webrtc:4865
R=juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1274013002 .

Cr-Commit-Position: refs/heads/master@{#9708}
2015-08-14 05:24:12 +00:00
f358aea7bf Fix WebRTC IP leaks.
WebRTC binds to individual NICs and listens for incoming Stun packets. Sending stun through this specific NIC binding could make OS route the packet differently hence exposing non-VPN public IP.

The fix here is
1. to bind to any address (0:0:0:0) instead. This way, the routing will be the same as how chrome/http is.
2. also, remove the any all 0s addresses which happens when we bind to all 0s.

BUG=4276
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39129004

Cr-Commit-Position: refs/heads/master@{#8418}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8418 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 18:44:14 +00:00
d3b453be17 Remove the incremental IP address behavior from virtualsocketserver
VirtualSocketServer, when binding to any address (all 0s), will assign a unique IP address by incrementing the IP address, resulted in 0.0.0.1. However, this breaks the testing of 4276 where we bind to all 0s and expect the local address should remain all 0s.

BUG=4276
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35189004

Cr-Commit-Position: refs/heads/master@{#8370}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8370 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-14 00:44:20 +00:00
ec499beaf5 Increase testclient timeout from 1 to 5 seconds
BUG=4182
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38839004

Cr-Commit-Position: refs/heads/master@{#8285}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8285 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-07 22:38:16 +00:00
c732a3e511 Re-enable allmost all base tests.
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/22989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 22:08:15 +00:00
fded02c164 base: disabled several base tests on Mac so that rtc_unittests can be turned back on
BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7240 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 13:10:10 +00:00
f048872e91 Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.

BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 18:00:26 +00:00
e9a604accd Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:15:48 +00:00
2c7d1b39b9 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 18:03:09 +00:00