Hello, I've got a problem with rtp handling by siemens c450 and similar. I experience a couple seconds of silence between early media and normal call (normal call's rtp is dropped by phone). This is caused by SSRC changing (even though marker bit is set). I have all relevant patches applied - it still happens on 1.4.21.1 and every version before that. Especially http://bugs.digium.com/view.php?id=12570 doesn't change anything, because call is p2p bridged.
Issue can be fixed by forcing use of the same ssrc in ast_raw_write and bridged rtp writes and my custom patch works, but I don't want to use it if it can be done in some other way. Is there a way to force treating outgoing rtp as one stream, instead of switching source after early media? Is there a way to do it without resigning from p2p bridging? Thanks for ideas, Stan _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
