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path: root/include/asterisk/frame.h
AgeCommit message (Expand)Author
2018-04-17bridge_softmix: Forward TEXT framesGeorge Joseph
2018-03-27res_rtp_asterisk: Add support for raising additional RTCP messages.Joshua Colp
2017-09-21bridge: Change participant SFU streams when source streams change.Joshua Colp
2017-07-19bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.Joshua Colp
2017-03-07core: Add stream topology changing primitives with tests.Joshua Colp
2017-02-24channel: Add ast_read_stream function for reading frames from all streams.Joshua Colp
2017-01-27Merge "media: Add experimental support for RTCP feedback."George Joseph
2017-01-23media: Add experimental support for RTCP feedback.Lorenzo Miniero
2017-01-17abstract/fixed/adpative jitter buffer: disallow frame re-insertsKevin Harwell
2014-10-03chan_pjsip: Fix deadlock when masquerading PJSIP channels.Richard Mudgett
2014-07-20media formats: re-architect handling of media for performance improvementsMatthew Jordan
2014-03-17Fix stuck channel in ARI through the introduction of synchronous bridge actions.Mark Michelson
2014-02-07chan_iax2: Block unnecessary control frames to/from the wire.Richard Mudgett
2013-08-21Set 14400 as the default max bit rate if T38MaxBitRate is not specifiedMatthew Jordan
2013-08-06ARI: Add recording controlsDavid M. Lee
2013-05-21Merge in the bridge_construction branch to make the system use the Bridging API.Richard Mudgett
2013-04-08Stasis application WebSocket supportDavid M. Lee
2013-01-22Add ControlPlayback manager actionMatthew Jordan
2012-07-20Add hangupcause translation supportKinsey Moore
2012-06-15Add HANGUPCAUSE hash support to IAX2Kinsey Moore
2012-05-14Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)Kinsey Moore
2011-09-16Merged revisions 336307 via svnmerge from Jonathan Rose
2011-09-09Merged revisions 335078 via svnmerge from Matthew Jordan
2011-06-23Merged revisions 324652 via svnmerge from David Vossel
2011-04-20Merged revisions 314417 via svnmerge from Richard Mudgett
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd a...David Vossel
2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
2010-12-01Merged revisions 296992 via svnmerge from Tilghman Lesher
2010-11-30Add a comment on why the reserved bit is reserved.Tilghman Lesher
2010-11-22Merged revisions 295866 via svnmerge from Richard Mudgett
2010-09-10Merged revisions 286189 via svnmerge from Terry Wilson
2010-06-17adds speex 16khz audio supportDavid Vossel
2010-06-16addition of G.719 pass-through supportDavid Vossel
2010-06-08Add SRTP support for AsteriskTerry Wilson
2010-06-02Generic Advice of Charge.Richard Mudgett
2010-05-17Enhancements to connected line and redirecting work.Mark Michelson
2010-04-21Added MixMonitorMute manager commandJulian Lyndon-Smith
2010-04-09Merge Call completion support into trunk.Mark Michelson
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capableKevin P. Fleming
2010-03-12Only change the RTP ssrc when we see that it has changedTerry Wilson
2010-03-02Merge res_fax and res_fax_spandsp.Matthew Nicholson
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
2009-10-21Merged revisions 224931 via svnmerge from Russell Bryant
2009-10-08Merged revisions 222878 via svnmerge from Russell Bryant
2009-07-23Rework of T.38 negotiation and UDPTL API to address interoperability problemsKevin P. Fleming
2009-07-09Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.Kevin P. Fleming
2009-07-08Merged revisions 205471 via svnmerge from David Vossel
2009-06-26Improve T.38 negotiation by exchanging session parameters between application...Joshua Colp
2009-06-16Enable applications to enable/disable digit and tone detection.Kevin P. Fleming
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