| f9945b2d1aa2d78b19987219ea872605167d7b5f |
|
15-Dec-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Only try to pair protocol matching candidates for creating connections.
If the local port and the remote candidate's protocols do not match,
do not even try to pair them.
This avoids printing out confusing logs like
"Attempt to change a remote candidate..." in p2ptransportchannel
when two remote candidates have the same port number but different
protocols.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516613002 .
Cr-Commit-Position: refs/heads/master@{#11034}
/external/webrtc/webrtc/p2p/base/stunport.h
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| e03cab94c1bac43f4d6c4775023a957f98ee8132 |
|
11-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
When running this code in chromium on a machine with IPv6 disabled, the RTC_DCHECK fails and in release build, it could leak to further crash in chromium's rtc_peer_connection_hanlder.cc.
Here is the right fix.
BUG=webrtc:5061
R=pthatcher@google.com
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1437933002 .
Cr-Commit-Position: refs/heads/master@{#10607}
/external/webrtc/webrtc/p2p/base/stunport.h
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| 9af97f89103d8f1f77b52a6ae77b8b7bcdc23f71 |
|
10-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
WebRTC should generate default private address even when adapter enumeration is disabled.
Introduce a DefaultAddressProvider such that rtc::Network can't access other part of NetworkManager.
This also removes the hack of generating the loopback address. The dependency has been removed by https://codereview.chromium.org/1417023003/
BUG=webrtc:5061
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1411253008 .
Cr-Commit-Position: refs/heads/master@{#10590}
/external/webrtc/webrtc/p2p/base/stunport.h
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| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/webrtc/p2p/base/stunport.h
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| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
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07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/p2p/base/stunport.h
|
| fe3bc9d5aeffed8bbfb34c330d8b991abd1a1aba |
|
20-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Relanding "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
Migrated from https://codereview.webrtc.org/1275703006/ which causes test failures for android. On android, loopback interface was used as local interface to generate candidates. Add a test case to make sure this won't be broken in the future.
Also observed some failures under content_browsertests in chromium.fyi bot but can't repro locally. Might just be temporary test issue.
BUG=webrtc:4517
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1299333003 .
Cr-Commit-Position: refs/heads/master@{#9746}
/external/webrtc/webrtc/p2p/base/stunport.h
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| 370c8848ad38d54457a960e0ebe94f8adf370e23 |
|
19-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
This reverts commit 0a2955f227666efd87b2a303a69c083ef801c528.
Revert "In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate."
This reverts commit ba9ab4cd8d2e8fbc068dc36b5e6f6331d7deeccf.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1288843003 .
Cr-Commit-Position: refs/heads/master@{#9729}
/external/webrtc/webrtc/p2p/base/stunport.h
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| 0a2955f227666efd87b2a303a69c083ef801c528 |
|
18-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied.
BUG=webrtc:4517
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1275703006 .
Cr-Commit-Position: refs/heads/master@{#9726}
/external/webrtc/webrtc/p2p/base/stunport.h
|
| 0ba1533fdbe4a098723da8262f1374d71c3a1806 |
|
10-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/stunport.h
|
| 332331fb01f8a316ac6d61cf4572478610fb3472 |
|
06-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use uint16s for port numbers in webrtc/p2p/base.
This is a necessary precursor to using uint16s for port numbers more
consistently in Chromium code.
This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override).
BUG=chromium:81439
TEST=none
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/stunport.h
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| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
|
28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/stunport.h
|
| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
|
18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/stunport.h
|
| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
|
15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/stunport.h
|