mercutioviz wrote:
definitely stop by IRC and talk real-time with others who've dealt with this kind of thing.
Thanks for the tip, but since this firewalls are a frequent issue with SIP, I think it'd be very useful if someone who has good experience with this wrote an article in the FS wiki for everyone to benefit. For instance, at this point, I don't know how data (SIP and RTP) flows between a remote SIP phone behind a NAT router, and a VoIP gateway located in the same LAN as a Freeswitch server both behind a NAT router. Do RTP packets go directly from the gateway to the remote SIP phone, or do they go through the FS server? That sort of thing :-) -- View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p230... Sent from the Freeswitch-users mailing list archive at Nabble.com.