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path: root/main/rtp_engine.c
AgeCommit message (Expand)Author
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
2013-02-12Do not allow native RTP bridging if packetization of media streams differs.Mark Michelson
2013-01-09* Found some more places to use ast_channel_lock_both().Richard Mudgett
2013-01-09Fix end condition in ast_rtp_lookup_mime_multiple2.David M. Lee
2012-11-07Multiple revisions 375993-375994Mark Michelson
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
2012-08-08Create the payload type if it does not exist when setting information based o...Joshua Colp
2012-08-07Payload and RTP code are must remain separate since in non-Asterisk format ca...Joshua Colp
2012-08-07Fix a bug uncovered by the test suite where the RTP payload number was not ge...Joshua Colp
2012-08-07Reduce memory consumption significantly for users of the RTP engine API by st...Joshua Colp
2012-07-20Add hangupcause translation supportKinsey Moore
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
2012-06-29Fix apparent copy and paste error where incorrect "glue" is used.Mark Michelson
2012-06-19Ensure that pvt cause information does not break native bridgingKinsey Moore
2012-06-15Multiple revisions 369001-369002Kevin P. Fleming
2012-05-24chan_sip: fix problem directmediapermit/deny uses the wrong addressJonathan Rose
2012-05-14Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)Kinsey Moore
2012-04-20Move debug message in ast_rtp_instance_early_bridge_make_compatible().Richard Mudgett
2012-04-20Use ast_channel_lock_both() where it was inlined before.Richard Mudgett
2012-03-22Kill off red blobs in most of main/*Kinsey Moore
2012-03-13Finalize ast_channel opaquificationTerry Wilson
2012-02-27Deprecated macro usage for connected line, redirecting, and CCSSKinsey Moore
2012-02-24Allow SRTP policies to be reloadedMatthew Jordan
2012-02-24Opaquification for ast_format structs in struct ast_channelTerry Wilson
2012-02-20ast_channel opaquification of pointers and integral typesTerry Wilson
2012-01-28Add 'L16-256' MIME subtype alias for slin16.Kevin P. Fleming
2012-01-16Add missing code to set direct RTP setup information during dialing.Joshua Colp
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
2011-12-29Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loopMatthew Jordan
2011-09-16Merged revisions 336307 via svnmerge from Jonathan Rose
2011-08-15Merged revisions 331894 via svnmerge from Paul Belanger
2011-07-19Merged revisions 328824 via svnmerge from Kinsey Moore
2011-07-07Adds pass-through support for codec CELT.David Vossel
2011-06-29Merged revisions 325537 via svnmerge from Matthew Nicholson
2011-06-14Merged revisions 323370 via svnmerge from Terry Wilson
2011-05-26Merged revisions 321042 via svnmerge from Terry Wilson
2011-05-03Merged revisions 316265 via svnmerge from Russell Bryant
2011-04-18Merged revisions 314017 via svnmerge from David Vossel
2011-03-11Merged revisions 310287 via svnmerge from Alec L Davis
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
2010-09-01Merged revisions 284477 via svnmerge from Terry Wilson
2010-07-29Merged revisions 280391 via svnmerge from Russell Bryant
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