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rtp_engine.c
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Author
2012-08-07
Reduce memory consumption significantly for users of the RTP engine API by st...
Joshua Colp
2012-07-20
Add hangupcause translation support
Kinsey Moore
2012-07-01
Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
Joshua Colp
2012-06-29
Fix apparent copy and paste error where incorrect "glue" is used.
Mark Michelson
2012-06-19
Ensure that pvt cause information does not break native bridging
Kinsey Moore
2012-06-15
Multiple revisions 369001-369002
Kevin P. Fleming
2012-05-24
chan_sip: fix problem directmediapermit/deny uses the wrong address
Jonathan Rose
2012-05-14
Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
Kinsey Moore
2012-04-20
Move debug message in ast_rtp_instance_early_bridge_make_compatible().
Richard Mudgett
2012-04-20
Use ast_channel_lock_both() where it was inlined before.
Richard Mudgett
2012-03-22
Kill off red blobs in most of main/*
Kinsey Moore
2012-03-13
Finalize ast_channel opaquification
Terry Wilson
2012-02-27
Deprecated macro usage for connected line, redirecting, and CCSS
Kinsey Moore
2012-02-24
Allow SRTP policies to be reloaded
Matthew Jordan
2012-02-24
Opaquification for ast_format structs in struct ast_channel
Terry Wilson
2012-02-20
ast_channel opaquification of pointers and integral types
Terry Wilson
2012-01-28
Add 'L16-256' MIME subtype alias for slin16.
Kevin P. Fleming
2012-01-16
Add missing code to set direct RTP setup information during dialing.
Joshua Colp
2012-01-09
Replace direct access to channel name with accessor functions
Terry Wilson
2011-12-29
Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop
Matthew Jordan
2011-09-16
Merged revisions 336307 via svnmerge from
Jonathan Rose
2011-08-15
Merged revisions 331894 via svnmerge from
Paul Belanger
2011-07-19
Merged revisions 328824 via svnmerge from
Kinsey Moore
2011-07-07
Adds pass-through support for codec CELT.
David Vossel
2011-06-29
Merged revisions 325537 via svnmerge from
Matthew Nicholson
2011-06-14
Merged revisions 323370 via svnmerge from
Terry Wilson
2011-05-26
Merged revisions 321042 via svnmerge from
Terry Wilson
2011-05-03
Merged revisions 316265 via svnmerge from
Russell Bryant
2011-04-18
Merged revisions 314017 via svnmerge from
David Vossel
2011-03-11
Merged revisions 310287 via svnmerge from
Alec L Davis
2011-02-22
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd a...
David Vossel
2011-02-03
Asterisk media architecture conversion - no more format bitfields
David Vossel
2010-12-20
Some scheduler API cleanup and improvements.
Russell Bryant
2010-11-03
Merged revisions 293803 via svnmerge from
Terry Wilson
2010-10-02
Merged revisions 289840 via svnmerge from
Jeff Peeler
2010-09-01
Merged revisions 284477 via svnmerge from
Terry Wilson
2010-07-29
Merged revisions 280391 via svnmerge from
Russell Bryant
2010-07-08
Add IPv6 to Asterisk.
Mark Michelson
2010-06-17
adds support for slin16 in sip
David Vossel
2010-06-17
adds speex 16khz audio support
David Vossel
2010-06-16
addition of G.719 pass-through support
David Vossel
2010-06-08
Add SRTP support for Asterisk
Terry Wilson
2010-06-07
Seems strange (and the code backs up) that if the max and min of a statistic ...
Tilghman Lesher
2010-05-17
Enhancements to connected line and redirecting work.
Mark Michelson
2010-03-12
Only change the RTP ssrc when we see that it has changed
Terry Wilson
2010-01-18
Fix an RTP instance allocation failure on Solaris.
Jason Parker
2009-12-01
More 32->64 bit codec conversions.
Tilghman Lesher
2009-11-04
Expand codec bitfield from 32 bits to 64 bits.
Tilghman Lesher
2009-10-01
Merged revisions 221776 via svnmerge from
Tilghman Lesher
2009-09-30
Use rtp properties instead of adding a callback
Terry Wilson
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