Quote:
Originally Posted by casch
I have set now in my ata:
Domainname/Realm: eu.voxalot.com:5060
Local Port: 5060
Proxy Server: eu.voxalot.com:5060
Register Server: eu.voxalot.com:5060
OutboundServer: eu.voxalot.com:5060
RTP Audio Port: 9000
RTP pkt Periode:20
pref. Codec: G.711A
RPORT: activated
STUN activated
(why this might harm?)
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If your NAT is too restricted it will not be compatible with STUN.
The STUN process is how your SIP user agent learns your external IP address and external port number. When port forwarding, port numbers remain unchanged as they traverse NAT, ie same port number on LAN and WAN. When port forwarding, if the SIP user agent can get your external IP address another way (through SIP registration or you enter it manually) then you don't need STUN.
Quote:
Originally Posted by casch
And (but?) under sip-status I see:
SIP NAT Typ:
Port Restricted NAT
(what does this mean?)
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See
Network address translation. STUN may or may not work for you, it depends on what type of NAT you have. If you have not already done so, you may want to experiment with the "Symmetric NAT" settings in your Voxalot account, and for outbound calls to PSTN you can try disabling "optimize audio path". Note this quote from Wikipedia; "This (NAT) terminology has been the source of much confusion, as it has proven inadequate at describing real-life NAT behavior."
If you are not port forwarding then in addition to using NAT you must use "NAT keep alive" packets to refresh the original NAT mapping which was created by SIP registration. If the "NAT keep alive" packet is sent after the original SIP registration mapping has expired (usually 180 seconds) a new mapping (with different port numbers) will be created which is useless.