6b9ab9204b9e0891cb0be2595a0d7ada92945cea |
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05-Jan-2016 |
honghaiz <honghaiz@webrtc.org> |
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}
/external/webrtc/webrtc/p2p/base/stunrequest.cc
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8597543ae845073ca32876bfaaff207017b6c0eb |
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12-Nov-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
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}
/external/webrtc/webrtc/p2p/base/stunrequest.cc
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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/stunrequest.cc
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3e95d3ef39c0e5e13d697dab9a8abea0e72feb4e |
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19-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
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}
/external/webrtc/webrtc/p2p/base/stunrequest.cc
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1cf6f8101ae9db517332783e99c98e14ff4c47e1 |
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15-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
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}
/external/webrtc/webrtc/p2p/base/stunrequest.cc
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ff689be3c0c59c1be29aaa0697aa0f762566d6c6 |
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12-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/stunrequest.cc
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fe672e3839004350403423b74764d42396a0edab |
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17-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
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
/external/webrtc/webrtc/p2p/base/stunrequest.cc
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0ba1533fdbe4a098723da8262f1374d71c3a1806 |
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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/stunrequest.cc
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269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
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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/stunrequest.cc
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28100cb38896fe298b6df11ffd31838d9faf5b8a |
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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/stunrequest.cc
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d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
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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/stunrequest.cc
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