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gerrit-public.fairphone.software / platform / external / webrtc / 45cc890560ef5aa678753cb07b59d0b299a8841a / p2p / base / relayport.cc
  1. 45cc890 Assorted logging pedantry by Jonas Olsson · 7 years ago
  2. 6c38cc7 Fix cpplint errors in p2p/ by Steve Anton · 7 years ago
  3. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  4. 1cf1b7d Fix clang style warnings in p2p/base/port.h and its subclasses by Steve Anton · 7 years ago
  5. 1c46a35 Try creating sockets again if network change occurs after bind failed. by deadbeef · 7 years ago
  6. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  7. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/p2p/base/relayport.cc]
  8. 5c3c104 Make Port (and subclasses) fully "Network"-based, instead of IP-based. by deadbeef · 7 years ago
  9. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  10. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  11. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  12. 996fc6b Don't crash if STUN error message is missing ERROR-CODE attribute. by deadbeef · 8 years ago
  13. f42cc9d Add MakeUnique from chromium and change StunMessage::AddAttribute to take a unique_ptr. by zstein · 8 years ago
  14. f137e97 Revert of Removing HTTPS and SOCKS proxy server code. (patchset #2 id:20001 of https://codereview.webrtc.org/2731673002/ ) by deadbeef · 8 years ago
  15. a1991c5 Removing HTTPS and SOCKS proxy server code. by deadbeef · 8 years ago
  16. 26d99c2 Add the URL attribute to cricket::Candiate. by zhihuang · 8 years ago
  17. cc99bc2 Change StunMessage::AddAttribute return type from bool to void. by nisse · 8 years ago
  18. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  19. 0483362 Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  20. d5236e2 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ ) by magjed · 8 years ago
  21. b0f04fd Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  22. 277b250 Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
  23. c309e0e Don't stop sending media on EWOULDBLOCK by skvlad · 8 years ago
  24. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  25. 36f50e8 Create a new connection if a candidate reuses an address by honghaiz · 8 years ago
  26. 1bffc1d Rename rtc::Time64 --> rtc::TimeMillis. by nisse · 9 years ago
  27. f1f8720 Split ByteBuffer into writer/reader objects. by jbauch · 9 years ago
  28. 34b11eb Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p. by honghaiz · 9 years ago
  29. 55674ff Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 9 years ago
  30. e5e0e57 Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) by tommi · 9 years ago
  31. 7307952 Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 9 years ago
  32. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  33. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  34. 3d564c1 Add instrumentation to track the IceEndpointType. by Guo-wei Shieh · 9 years ago
  35. 1cf6f81 Add logging for sending and receiving STUN binding requests and TURN requests and responses. by Peter Thatcher · 9 years ago
  36. ff689be Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. by andresp@webrtc.org · 10 years ago
  37. 332331f Use uint16s for port numbers in webrtc/p2p/base. by pkasting@chromium.org · 10 years ago
  38. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  39. 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  40. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago