Don't do reinvite if both parties talk diffrent codecs
[asterisk/asterisk.git] / include / asterisk / rtp.h
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