[Asterisk-Users] Problem In RTC Client When Used With Asterisk

[Danish Samad]

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