376e1235c7b602e86afe9f36eb81289e42643718 |
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25-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Destroy a Connection if a CreatePermission request fails. This means that if a TURN server denies permission for an unreachable address, we'll no longer ping it fruitlessly. BUG=webrtc:4917 Review URL: https://codereview.webrtc.org/1415313004 Cr-Commit-Position: refs/heads/master@{#10789}
/external/webrtc/webrtc/p2p/base/turnserver.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/turnserver.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/turnserver.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/turnserver.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/turnserver.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/turnserver.cc
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