I'm interested to hear how folks are handling NAT SIP routing issues "in the
wild" for commercial use.
Are you using a commerical SIP-aware NAT router solution? If so, what?
Are you using a software SIP-proxy like SER or siproxd? If so, which?
Do you set everything to "canreinvite=no" in sip.conf?
Any comments about real-world implementations would be welcome.
We handle it via SER with its rtpproxy/nathelper modules. Our
configuration can detect automatically if the mediastream should be
handled by our servers or if the endpoints successfully used STUN and
can communicate directly.
--
Andres
Network Admin
http://www.telesip.net
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