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rtp_engine.h
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2016-04-05
Merge "res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS" into 13
Joshua Colp
2016-03-29
res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS
Jacek Konieczny
2016-03-28
res_rtp_asterisk: Fix packet stats on bridged connection
George Joseph
2016-02-23
rtp_engine.h: Remove extraneous semicolons.
Richard Mudgett
2015-07-30
rtp_engine.h: No sense allowing payload types larger than RFC allows.
Richard Mudgett
2015-07-30
rtp_engine.h: Misc comment fixes.
Richard Mudgett
2015-07-24
pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
Joshua Colp
2015-07-20
res_pjsip: Add rtp_keepalive endpoint option.
Mark Michelson
2014-12-09
Direct Media calls within private network sometimes get one way audio
Kevin Harwell
2014-09-16
res_rtp_asterisk: Fix a myriad of TURN client issues.
Joshua Colp
2014-07-20
media formats: re-architect handling of media for performance improvements
Matthew Jordan
2014-06-30
Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/...
Joshua Colp
2014-04-15
chan_sip.c: Fix channel staging assertion failure.
Richard Mudgett
2014-03-20
assigned-uniqueids: Miscellaneous cleanup and fixes.
Richard Mudgett
2013-11-01
chan_sip: Fix RTCP port for SRFLX ICE candidates
Kinsey Moore
2013-10-26
rtp_engine: fix rtp payloads copy and improve argument names
Scott Griepentrog
2013-07-05
Refactor RTCP events over to Stasis; associate with channels
Matthew Jordan
2013-05-21
Merge in the bridge_construction branch to make the system use the Bridging API.
Richard Mudgett
2013-03-07
Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source
Matthew Jordan
2012-10-04
Add support for applying direct media ACLs between differing channel technolo...
Joshua Colp
2012-09-20
Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
Joshua Colp
2012-09-05
Re-fix sending unnegotiated payloads during a P2P RTP bridge.
Mark Michelson
2012-08-30
Clean up doxygen warnings
Matthew Jordan
2012-08-07
Payload and RTP code are must remain separate since in non-Asterisk format ca...
Joshua Colp
2012-08-07
Reduce memory consumption significantly for users of the RTP engine API by st...
Joshua Colp
2012-07-18
Fix a crash occurring as a result of excess stack usage.
Joshua Colp
2012-07-01
Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
Joshua Colp
2012-05-24
chan_sip: fix problem directmediapermit/deny uses the wrong address
Jonathan Rose
2012-02-24
Allow SRTP policies to be reloaded
Matthew Jordan
2011-07-21
Merged revisions 329257 via svnmerge from
Russell Bryant
2011-07-19
Merged revisions 328824 via svnmerge from
Kinsey Moore
2011-06-14
Merged revisions 323370 via svnmerge from
Terry Wilson
2011-04-18
Merged revisions 314017 via svnmerge from
David Vossel
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
2010-07-08
Add IPv6 to Asterisk.
Mark Michelson
2010-06-08
Fix some doxygen warnings.
Leif Madsen
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
2010-04-09
Merge Call completion support into trunk.
Mark Michelson
2010-03-12
Only change the RTP ssrc when we see that it has changed
Terry Wilson
2010-03-05
Fix up the ast_rtp_property enum.
Russell Bryant
2009-12-21
Change all refererences to 1.6.3 to be 1.8, since that will be the next featu...
Kevin P. Fleming
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
2009-09-30
Use rtp properties instead of adding a callback
Terry Wilson
2009-09-30
Merged revisions 221086 via svnmerge from
Terry Wilson
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