git migration: Refactor the ASTERISK_FILE_VERSION macro
[asterisk/asterisk.git] / main / rtp_engine.c
2015-04-13 Matt Jordangit migration: Refactor the ASTERISK_FILE_VERSION macro
2015-03-26 Corey FarrellReplace most uses of ast_register_atexit with ast_regis...
2015-03-23 Matthew JordanFix compilations errors on 64-bit OpenBSD systems
2015-01-26 David M. LeeVarious fixes for OS X
2015-01-21 Matthew Jordanmain/rtp_engine: Format NTP timestamps as unsigned...
2015-01-07 Kinsey MooreFix dev-mode build on recent gcc
2014-12-31 Scott Griepentrogrtp_engine: keep payload types in correct range
2014-12-09 Kevin HarwellDirect Media calls within private network sometimes...
2014-12-01 Joshua Colprtp_engine: Add support for transporting signed linear...
2014-11-13 Matthew Jordanmain/rtp_engine: Fix crash when processing more than...
2014-09-06 Matthew Jordanmain/rtp_engine: Format NTP timestamps as unsigned...
2014-08-06 Kinsey MooreStasis: Allow message types to be blocked
2014-07-31 Matthew Jordanres_hep_rtcp: Add module that sends RTCP information...
2014-07-20 Matthew Jordanmedia formats: re-architect handling of media for perfo...
2014-06-30 Joshua ColpRecorded merge of revisions 417677 from svn.asterisk...
2014-05-13 Walter Doekesrtp: Fix case typo in H263+ mime.
2014-05-09 Kinsey MooreAllow Asterisk to compile under GCC 4.10
2014-04-15 Richard MudgettEliminate some more unnecessary RAII_VAR() uses.
2014-04-15 Richard Mudgettchan_sip.c: Fix channel staging assertion failure.
2014-02-21 Kevin Harwellrtp_engine: Dynamic payload change in rtp mapping not...
2014-02-21 Kevin Harwellrtp_engine: Output mixup in ${CHANNEL(rtpqos,audio...
2014-01-28 Scott Griepentrogrtp_engine: improved handling of get_rtp_info failure
2013-12-17 Rusty NewtonSeveral components: fixing Typos in comments and code...
2013-11-22 Kinsey MooreARI: Don't leak implementation details
2013-10-26 Scott Griepentrogrtp_engine: fix rtp payloads copy and improve argument...
2013-10-03 Mark MichelsonCache string values of formats on ast_format_cap()...
2013-08-23 Matthew JordanAdd pass through support for Opus and VP8; Opus format...
2013-08-16 Richard MudgettDoxygen comment tweaks.
2013-08-02 Mark MichelsonGet rid of ast_bridged_channel() and the bridged_channe...
2013-08-01 Kinsey MooreFix documentation replication issues
2013-07-05 Matthew JordanRefactor RTCP events over to Stasis; associate with...
2013-05-21 Richard MudgettMerge in the bridge_construction branch to make the...
2013-03-07 Matthew JordanAdd a 'secret' probation strictrtp mode to handle delay...
2013-02-12 Mark MichelsonDo not allow native RTP bridging if packetization of...
2013-01-09 Automerge scriptMerged revisions 378783,378789-378790 via svnmerge...
2013-01-09 Richard Mudgett* Found some more places to use ast_channel_lock_both().
2013-01-09 David M. LeeFix end condition in ast_rtp_lookup_mime_multiple2.
2012-11-07 Mark MichelsonMultiple revisions 375993-375994
2012-10-04 Joshua ColpAdd support for applying direct media ACLs between...
2012-09-20 Joshua ColpAdd support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
2012-09-05 Mark MichelsonRe-fix sending unnegotiated payloads during a P2P RTP...
2012-08-08 Joshua ColpCreate the payload type if it does not exist when setti...
2012-08-07 Joshua ColpPayload and RTP code are must remain separate since...
2012-08-07 Joshua ColpFix a bug uncovered by the test suite where the RTP...
2012-08-07 Joshua ColpReduce memory consumption significantly for users of...
2012-07-20 Kinsey MooreAdd hangupcause translation support
2012-07-01 Joshua ColpAdd support for ICE/STUN/TURN in res_rtp_asterisk and...
2012-06-29 Mark MichelsonFix apparent copy and paste error where incorrect ...
2012-06-19 Kinsey MooreEnsure that pvt cause information does not break native...
2012-06-15 Kevin P. FlemingMultiple revisions 369001-369002
2012-05-24 Jonathan Rosechan_sip: fix problem directmediapermit/deny uses the...
2012-05-14 Kinsey MooreCommit framework for HANGUPCAUSE (replacement for SIP_C...
2012-04-20 Richard MudgettMove debug message in ast_rtp_instance_early_bridge_mak...
2012-04-20 Richard MudgettUse ast_channel_lock_both() where it was inlined before.
2012-03-22 Kinsey MooreKill off red blobs in most of main/*
2012-03-13 Terry WilsonFinalize ast_channel opaquification
2012-02-27 Kinsey MooreDeprecated macro usage for connected line, redirecting...
2012-02-24 Matthew JordanAllow SRTP policies to be reloaded
2012-02-24 Terry WilsonOpaquification for ast_format structs in struct ast_channel
2012-02-20 Terry Wilsonast_channel opaquification of pointers and integral...
2012-01-28 Kevin P. FlemingAdd 'L16-256' MIME subtype alias for slin16.
2012-01-16 Joshua ColpAdd missing code to set direct RTP setup information...
2012-01-09 Terry WilsonReplace direct access to channel name with accessor...
2011-12-29 Matthew JordanHandle AST_CONTROL_UPDATE_RTP_PEER frames in local...
2011-09-16 Jonathan RoseMerged revisions 336307 via svnmerge from
2011-08-15 Paul BelangerMerged revisions 331894 via svnmerge from
2011-07-19 Kinsey MooreMerged revisions 328824 via svnmerge from
2011-07-07 David VosselAdds pass-through support for codec CELT.
2011-06-29 Matthew NicholsonMerged revisions 325537 via svnmerge from
2011-06-14 Terry WilsonMerged revisions 323370 via svnmerge from
2011-05-26 Terry WilsonMerged revisions 321042 via svnmerge from
2011-05-03 Russell BryantMerged revisions 316265 via svnmerge from
2011-04-18 David VosselMerged revisions 314017 via svnmerge from
2011-03-11 Alec L DavisMerged revisions 310287 via svnmerge from
2011-02-22 David VosselMedia Project Phase2: SILK 8khz-24khz, SLINEAR 8khz...
2011-02-03 David VosselAsterisk media architecture conversion - no more format...
2010-12-20 Russell BryantSome scheduler API cleanup and improvements.
2010-11-03 Terry WilsonMerged revisions 293803 via svnmerge from
2010-10-02 Jeff PeelerMerged revisions 289840 via svnmerge from
2010-09-01 Terry WilsonMerged revisions 284477 via svnmerge from
2010-07-29 Russell BryantMerged revisions 280391 via svnmerge from
2010-07-08 Mark MichelsonAdd IPv6 to Asterisk.
2010-06-17 David Vosseladds support for slin16 in sip
2010-06-17 David Vosseladds speex 16khz audio support
2010-06-16 David Vosseladdition of G.719 pass-through support
2010-06-08 Terry WilsonAdd SRTP support for Asterisk
2010-06-07 Tilghman LesherSeems strange (and the code backs up) that if the max...
2010-05-17 Mark MichelsonEnhancements to connected line and redirecting work.
2010-03-12 Terry WilsonOnly change the RTP ssrc when we see that it has changed
2010-01-18 Jason ParkerFix an RTP instance allocation failure on Solaris.
2009-12-01 Tilghman LesherMore 32->64 bit codec conversions.
2009-11-04 Tilghman LesherExpand codec bitfield from 32 bits to 64 bits.
2009-10-01 Tilghman LesherMerged revisions 221776 via svnmerge from
2009-09-30 Terry WilsonUse rtp properties instead of adding a callback
2009-09-30 Terry WilsonMerged revisions 221086 via svnmerge from
2009-09-29 Mark MichelsonGet rid of annoying and cryptic debug messages.
2009-08-16 Joshua ColpAdd two more API calls for getting the current glue...
2009-08-13 Joshua ColpAdd an API call for retrieving the engine in use by...
2009-07-23 Kevin P. FlemingRework of T.38 negotiation and UDPTL API to address...
2009-06-26 Joshua ColpImprove T.38 negotiation by exchanging session paramete...
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