Hi, On 3/4/10 9:31 PM, James Lamanna wrote: > Hi, > I'm trying to use OpenSips as a registration server for Asterisk > (assuming we can get presence working ok). > Do I need to setup and use MediaProxy (or similar)? Or is the > nathelper stuff good enough? > I've made test calls from phones behind NAT to opensips to asterisk > and I haven't experienced any > one-way audio problems. > Also, the phones are not allowed to reinvite, because I need to keep > Asterisk in the media path.
If you want Asterisk to stay in the media path then you don't need MediaProxy. By setting nat=yes for a SIP peer Asterisk uses the so called COMEDIA mode, that is, it waits to receive RTP and sends RTP to where it received it from, not trusting the SDP. Regards, -- Saúl Ibarra Corretgé AG Projects _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
