Commit Verlauf

Autor SHA1 Nachricht Datum
  Joshua Colp 96b7d10d0c res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't. vor 10 Jahren
  Joshua Colp 6dd8e5300d res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'. vor 10 Jahren
  Kevin Harwell 9869172e3a res_rtp_asterisk: Crash if no candidates received for component vor 10 Jahren
  Joshua Colp 9b4763dbe5 res_rtp_asterisk: Allow only UDP ICE candidates. vor 10 Jahren
  Joshua Colp 106b3ebbd2 res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP. vor 10 Jahren
  Joshua Colp 4b7f9cc2d6 res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered. vor 10 Jahren
  Joshua Colp e87013df29 res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. vor 10 Jahren
  Joshua Colp d483b133fe res_rtp_asterisk: Fix building when pjproject is not used. vor 10 Jahren
  Joshua Colp 3b6b094ca8 res_rtp_asterisk: Fix a myriad of TURN client issues. vor 10 Jahren
  Matthew Jordan cfcaa96fc6 res_hep_rtcp: Add module that sends RTCP information to a Homer Server vor 10 Jahren
  Mark Michelson 9b08352d97 Add module support level to ast_module_info structure. Print it in CLI "module show" . vor 10 Jahren
  Matthew Jordan 720c59c321 media formats: re-architect handling of media for performance improvements vor 10 Jahren
  Matthew Jordan d740e30bb5 res_rtp_asterisk: Fix undefined function when PJPROJECT is not installed vor 10 Jahren
  Joshua Colp 8a9991d4d8 res_rtp_asterisk: Don't leak memory or reset state if DTLS configuration is set multiple times. vor 10 Jahren
  Joshua Colp 4eee582865 Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 vor 10 Jahren
  Joshua Colp 9f84578290 res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0. vor 10 Jahren
  Kinsey Moore 43b85a2bbe Allow Asterisk to compile under GCC 4.10 vor 10 Jahren
  Matthew Jordan 16e7064f3c res_rtp_asterisk: Add support for DTLS handshake retransmissions vor 10 Jahren
  Jonathan Rose ae9a9d3a2c res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE vor 10 Jahren
  Jonathan Rose 895a113511 Multiple revisions 409129-409130 vor 10 Jahren
  Corey Farrell debb75f735 res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions vor 10 Jahren
  Kevin Harwell 112aa205f6 res_rtp_asterisk: Fails to resume WebRTC call from hold vor 10 Jahren
  Kinsey Moore b5e28ad8f1 chan_sip: Fix RTCP port for SRFLX ICE candidates vor 11 Jahren
  Jonathan Rose 0c2d008462 res_rtp_asterisk: Address jittery DTMF events in RTP streams vor 11 Jahren
  Matthew Jordan a81c2e392f res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change vor 11 Jahren
  Kinsey Moore be9e03a18e Fix STUN crash when using IPv6 any address vor 11 Jahren
  Matthew Jordan 4aac8124d8 res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports vor 11 Jahren
  Richard Mudgett 0c9611a6c4 res_rtp_asterisk: Fix ref leaks in ast_rtcp_read(). vor 11 Jahren
  David M. Lee 9f1c500b79 res_pjsip: Forward PJSIP logging to Asterisk logging vor 11 Jahren
  Matthew Jordan 76b04fcfbb Add pass through support for Opus and VP8; Opus format attribute negotiation vor 11 Jahren