I've been doing some experiments with app_meetme, and only have SIP phones here to play with. I have been finding issues with audio delay that I think may be to do with the use of pseudo channels to conference non-Zap channels.
The easiest way to demonstrate it is first of all to make a pair of calls to an extension that calls MeetMe(2222|). Speaking into both phones and listening to them both gives an audio delay of about 300-400ms. Then repeat the experiment using MeetMe(2222|q). This time the audio comes back almost instantaneously. I am suspecting that the problem is something to do with the conf_play() of the enter and leave sounds. My guess is that by writing that raw data into the pseudo device fd, it causes a backlog that never drains, because the device is only getting emptied at the same rate as the conference is filling it. The delay does seem to be of approximately the same length as the enter sound. I don't know whether the same issue applies to direct Zap channels or not. Any comments? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
