Commit Graph

15 Commits

Author SHA1 Message Date
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
fa301809b6 Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.

TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1274273005

Cr-Commit-Position: refs/heads/master@{#9698}
2015-08-11 11:13:00 +00:00
3449faa553 Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1263663002 .

Cr-Commit-Position: refs/heads/master@{#9692}
2015-08-10 19:22:59 +00:00
f421bdc68d Fix an NPE when creating TurnPort with a NULL socket.
BUG=4827

Review URL: https://codereview.webrtc.org/1241943002

Cr-Commit-Position: refs/heads/master@{#9601}
2015-07-17 23:22:02 +00:00
c5d0d95fd8 Ensuring that UDP TURN servers are always used as STUN servers.
This was already working in most cases, but not for some corner cases:
* If the PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is not set
* If both a STUN server and TURN server are configured

I added unit tests for these cases, and centralized the code that gets
STUN server addresses in order to fix these and any related issues.

BUG=webrtc:4215

Review URL: https://codereview.webrtc.org/1215713003

Cr-Commit-Position: refs/heads/master@{#9596}
2015-07-16 17:22:28 +00:00
894ad94302 Fix occurrences of const typed declaration without initialization
This fixes compilation errors as the following:

error: constructor must explicitly initialize the const member

BUG=506663
R=aluebs@webrtc.org, tommi@webrtc.org

Signed-off-by: Eduardo Lima (Etrunko) <eduardo.lima@intel.com>

Review URL: https://codereview.webrtc.org/1222233002

Cr-Commit-Position: refs/heads/master@{#9538}
2015-07-03 15:34:40 +00:00
d7e5c44e94 STUN allocation should not be disabled when using shared port and TURN servers are provided.
BUG=
R=juberti@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48229004

Cr-Commit-Position: refs/heads/master@{#9091}
2015-04-27 18:46:58 +00:00
931e0cf4b1 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

Committed: https://code.google.com/p/webrtc/source/detail?r=8418

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

Cr-Commit-Position: refs/heads/master@{#8419}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8419 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 19:10:22 +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
8cf9bdb3fa Remove USE_WEBRTC_DEV_BRANCH.
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.

R=bjornv@webrtc.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39849004

Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:17:12 +00:00
7e5b380437 Fix a crash in AllocationSequence.
Internal bug 19074679.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8130 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 21:28:39 +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