asterisk users - Oct 2004 - Problem In RTC Client When Used With Asterisk

Hi 
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in many
calls, Caller can hear the voice of the receiving side
but the receiver cant be able hear the caller for
about 5 to 10 seconds, conversation will become two
way after 5 - 10 seconds but this problem is a big
hurdle in proper establishment of a call

Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance


		
_______________________________
Do you Yahoo!?
Declare Yourself - Register online to vote today!
http://vote.yahoo.com

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