Fix call timeouts with rtp bridge etc (bug #5252)
[asterisk/asterisk.git] / include / asterisk / rtp.h
2005-10-13 Mark SpencerFix call timeouts with rtp bridge etc (bug #5252)
2005-10-12 Russell Bryantfix some formatting and add some comments (issue #5403)
2005-09-09 Kevin P. Flemingallow users of RTP to know when the peer endpoint is...
2005-08-30 Kevin P. Flemingmajor header file cleanup: license, copyrights, descrip...
2005-04-21 Kevin P. Fleminguse double-quotes instead of angle-brackets for non...
2005-04-05 Kevin P. Flemingreport non-codec capabilities in 'sip debug' properly...
2005-03-04 Mark SpencerRework channel structure to eliminate "pvt" portion...
2005-01-15 Mark SpencerRepair // comments to /* */ comments (bug #3347)
2005-01-07 Mark SpencerSupport CNG transmission when on hold (bug #2904)
2004-12-28 Mark SpencerPermit RTP to be reset
2004-08-26 Mark SpencerRepair offer/answer model (bug #2293), initial CNG...
2004-07-08 Mark SpencerExtend bindaddr to RTP connections on SIP (bug #1989...
2004-05-27 Mark SpencerMake RTP handle codecs (first pass)
2003-11-15 Martin PyckoDon't do reinvite if both parties talk diffrent codecs
2003-06-28 Mark SpencerAdd SIP/RTP video support, video enable app_echo, start...
2003-05-16 Mark SpencerMake RTP ports configurable
2003-05-02 Mark SpencerShow uptime
2003-03-20 Mark SpencerDon't destory rtp until destroy, use rtp_stop instead
2003-03-13 Matteo BrancaleoniThu Mar 13 07:00:01 CET 2003
2003-03-12 Matteo BrancaleoniWed Mar 12 07:00:01 CET 2003
2003-03-07 Matteo BrancaleoniFri Mar 7 07:00:00 CET 2003
2003-02-16 Matteo BrancaleoniSun Feb 16 07:00:01 CET 2003
2003-02-12 Matteo Brancaleonimer feb 12 14:56:57 CET 2003
2003-01-17 Mark SpencerVersion 0.3.0 from FTP
2002-07-31 Mark SpencerVersion 0.2.0 from FTP
2002-06-16 Mark SpencerVersion 0.1.12 from FTP