[EMAIL PROTECTED] wrote:
It's not a fault of the STUN server.Hi Karl, I'm suffering with the problem you outlined in (a) regardless of a STUN Server being used.
Is their anyway around this?
Yes, with a little patience there will be a way around this. We are close to releasing STUN support for Asterisk SIP for beta testing soon and it's running now in development versions. The changes in SIP to make this work properly are pretty pervasive, as every time a RTP port gets opened, STUN needs to perform a binding request across the NAT.
Then this information needs to be placed into the request headers and session description protocol
This also means, unfortunately, that there will be additional latency on call initiation and this needs to be kept and tuned to a minimum, but is never avoidable.
The whole implementation makes the SIP channel auto configuring, it discovers its main interface (or you can still
tell it to bind to eth0, for example), it automatically configures the localnet/mask feature, the externip, it tells you the type of NAT you have and what its properties are w/r/t hair pinning and port preserving character (primary and secondary), and other goodies.
Cheers, Sahil
STUN (RFC-3489) is an UNSAF type network protocol (see RFC 3424) that is used to discover UDP address and port bindings across network address translators.
(a) Currently Asterisk only supports static configuration of the external IP address of a NAT. You need to discover it manually by other means and configure SIP channel. This method fails for certain types of NATs that don't preserve port mapping from inside to outside across the NAT. i.e. if you are originating a request from ipaddress:5060, the NAT may map it to anotheraddress:15345 and this mapping may not be predictable, therefore asterisk cannot send proper SIP headers and will fail.
(b) Some ISP providers who use dynamic IPs will force your NAT router
to refresh its IP address periodically
and assign you a different one, at which time it would be nice to
automatically recognize that without having to
shut down your Asterisk and restarting.
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