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VoIP NAT: Difference between revisions
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→Tips for VoIP through NAT
(Mention configuring the public address, or using a STUN server to obtain it) |
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*SIP and NAT requires the call server, NAT device and phone to all play nicely and can still mean problems. There are a few specific cases we have tested and found reliable, but we cannot guarantee it will work in all cases or without some specific configuration settings
==Tips and recommendations for VoIP through NAT==
If you do want to use NAT, then here are some tips if you are struggling to get calls working:
*'''Have your broadband with us.''' - we provide public IPs on all our connections.
*'''Disable UPnP''' on routers
*'''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. - 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
*
*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
* '''Avoid ISPs that do CGNAT''' - that is NAT within their own network. - Having multiple levels of NAT between your handset and our service can cause problems in practice. Ask your ISP to provided you with a public IP address and no CGNAT.
If you have
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.
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