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path: root/include/asterisk/rtp_engine.h
AgeCommit message (Expand)Author
2017-12-14res_rtp_asterisk.c: Disable packet flood detection for video streams.Richard Mudgett
2017-08-04res_rtp_asterisk: Make P2P bridge Asymmetric codec awareTorrey Searle
2017-06-27bridge_native_rtp: Keep rtp instance refs on bridge_channelGeorge Joseph
2017-03-15Add rtcp-mux supportMark Michelson
2017-01-12res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt)Aaron An
2016-11-02rtp_engine: Allow more than 32 dynamic payload types.Alexander Traud
2016-04-05Merge "res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS" into 13Joshua Colp
2016-03-29res_rtp_asterisk: Use separate SRTP session for RTCP with DTLSJacek Konieczny
2016-03-28res_rtp_asterisk: Fix packet stats on bridged connectionGeorge Joseph
2016-02-23rtp_engine.h: Remove extraneous semicolons.Richard Mudgett
2015-07-30rtp_engine.h: No sense allowing payload types larger than RFC allows.Richard Mudgett
2015-07-30rtp_engine.h: Misc comment fixes.Richard Mudgett
2015-07-24pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.Joshua Colp
2015-07-20res_pjsip: Add rtp_keepalive endpoint option.Mark Michelson
2014-12-09Direct Media calls within private network sometimes get one way audioKevin Harwell
2014-09-16res_rtp_asterisk: Fix a myriad of TURN client issues.Joshua Colp
2014-07-20media formats: re-architect handling of media for performance improvementsMatthew Jordan
2014-06-30Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/...Joshua Colp
2014-04-15chan_sip.c: Fix channel staging assertion failure.Richard Mudgett
2014-03-20assigned-uniqueids: Miscellaneous cleanup and fixes.Richard Mudgett
2013-11-01chan_sip: Fix RTCP port for SRFLX ICE candidatesKinsey Moore
2013-10-26rtp_engine: fix rtp payloads copy and improve argument namesScott Griepentrog
2013-07-05Refactor RTCP events over to Stasis; associate with channelsMatthew Jordan
2013-05-21Merge in the bridge_construction branch to make the system use the Bridging API.Richard Mudgett
2013-03-07Add a 'secret' probation strictrtp mode to handle delayed changes in RTP sourceMatthew Jordan
2012-10-04Add support for applying direct media ACLs between differing channel technolo...Joshua Colp
2012-09-20Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.Joshua Colp
2012-09-05Re-fix sending unnegotiated payloads during a P2P RTP bridge.Mark Michelson
2012-08-30Clean up doxygen warningsMatthew Jordan
2012-08-07Payload and RTP code are must remain separate since in non-Asterisk format ca...Joshua Colp
2012-08-07Reduce memory consumption significantly for users of the RTP engine API by st...Joshua Colp
2012-07-18Fix a crash occurring as a result of excess stack usage.Joshua Colp
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
2012-05-24chan_sip: fix problem directmediapermit/deny uses the wrong addressJonathan Rose
2012-02-24Allow SRTP policies to be reloadedMatthew Jordan
2011-07-21Merged revisions 329257 via svnmerge from Russell Bryant
2011-07-19Merged revisions 328824 via svnmerge from Kinsey Moore
2011-06-14Merged revisions 323370 via svnmerge from Terry Wilson
2011-04-18Merged revisions 314017 via svnmerge from David Vossel
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd a...David Vossel
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
2010-12-20Some scheduler API cleanup and improvements.Russell Bryant
2010-11-03Merged revisions 293803 via svnmerge from Terry Wilson
2010-10-02Merged revisions 289840 via svnmerge from Jeff Peeler
2010-07-08Add IPv6 to Asterisk.Mark Michelson
2010-06-08Fix some doxygen warnings.Leif Madsen
2010-06-08Add SRTP support for AsteriskTerry Wilson
2010-06-07Seems strange (and the code backs up) that if the max and min of a statistic ...Tilghman Lesher
2010-04-09Merge Call completion support into trunk.Mark Michelson
2010-03-12Only change the RTP ssrc when we see that it has changedTerry Wilson
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