VoIP NAT: Difference between revisions

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If you do want to use NAT, then here are some tips if you are struggling:

*Disable UPnP on routers

*'''Disable UPnP''' on routers

*Disable SIP ALG on the router (or try enabling) See [[Disable SIP ALG]]

*'''Disable SIP ALG''' on the router (or try enabling) See [[Disable SIP ALG]] - ALG can do funky things!

*Reduce the registration 'expiry' - try 10 minutes (600 seconds) the default on devices us usually an hour.

*Reduce the '''registration 'expiry'''' - try '''10 minutes''' (600 seconds) the default on devices us usually an hour. - helps keeps the registration active

*Set '''SIP Keepalive''' (if the phone has this option) to '''30 seconds''' - this helps keep the NAT session live on the router

*If the VoIP phone has a config setting to enter your external/public IP address, then enter the address of your router's WAN port

*~~Disable~~ Stun settings on the VoIP phone, or ~~try~~ ~~enabling~~ it - stun.aa.net.uk can be used as the server name

*'''Enable Stun''' settings on the VoIP phone, or disable stun it - stun.aa.net.uk can be used as the server name

*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]]

*Enter firewall rules on the router to allow UDP traffic from our VoIP servers to your VoIP phone. See [[VoIP Firewall]] - usually not needed for NAT, but needed if you have public IPs

If you have 2 similar VoIP phones behind a router using NAT, you ~~will~~ have to change the port settings on one phone.

If you have two similar VoIP phones behind a router using NAT, you may have to change the port settings on one phone.

The local SIP port (often 5060) and the local RTP port(s) can't be the same on the 2 phones - if they are the same, you'll

get weirdness on incoming calls.