HI, I have used RTC with other SIP Proxies like SER and party sip and it works fine, never tested it with asterisk though. Basically Asterisk initiallly proxies RTP through itself and then sends reinvites to both endpoints to make RTP flow directly between the two gateways. Asterisk does have problems with the packetization perid values. It might be the case that the RTC endpoints use a different packetization period as compared to asterisk and it is only when the RTP goes direct, the endpoints start using the same packetization.
Whatever the problem maybe, I would suggest capturing SIP and media packets on both server and client side and analyzing them. You can use ethereal (www.ethereal.com) for this purpose, it is an extremely useful opensource network analyzer. Hope this helps, Danish _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
