In article <[EMAIL PROTECTED]>, Steven Langley <[EMAIL PROTECTED]> wrote: > > I am using IAX2 softphones dialing into meetme conferences. I also have > jitterbuffer=yes, with typical jitterbuffer settings. The problem I am > having is that as soon as there is a delay from a participant, then the > delay continues until the participant hangs up and dials in again. When > dialing in again the delay seems to go. > > It seems to me as though as soon as the server registers a delay from a > participant, then it causes delay on all further packets from that > participant. > > Does anyone have any ideas what the problem could be?
Yes, there are a few possibilities. Firstly, are you using ztdummy for timing? Which kernel version? If 2.6, have you ensured that USE_RTC is correctly defined in ztdummy.c? Look in bugs.digium.com at bug IDs 3599 and 4252 - they might be relevant. Yesterday I found another mechanism which could give rise to both a delay and broken audio - I found it with OH323 channels, but it might possibly arise on other channel types too. It concerns a backlog building up in the channel driver and never being drained by meetme because of blocking in the pseudo-device when trying to write the contents of a large frame. In app_meetme.c, try replacing this: careful_write(fd, f->data, f->datalen); with this: if (f->datalen <= CONF_SIZE) { careful_write(fd, f->data, f->datalen); } else { ast_log(LOG_WARNING, "Discarding large frame (%d bytes) from channel %s\n", f->datalen, chan->name); } and see if it helps. I haven't yet submitted the above change to mantis. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users