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VoIP NAT

77 bytes added, 16:02, 8 November 2020
Move the "purpose of the page" to the top
You have probably been directed to this page because you are having trouble with Voice over IP services (using SIP) when using a router or firewall which does some sort of Network Address Translation (NAT). This page tries to explain some of the issues, and why it is often a problem.
 
NAT is not officially supported, but generally can be made to work. Due to the nature of NAT, and the numerous implementation and 'fixes' and 'bodges' in routers, it can be tricky to get working. Lots of the phones that we've tested do just work without the need for an ALG, Stun, Port Forwarding etc., but other network equipment (i.e. the router) may get in the way.
 
 
When your VoIP phone places a call, a message is sent on a control channel - this will conventionally be to the VoIP server's IP address on port 5060. When the call is answered, the two ends of the link have a conversation to agree on how to do the audio link (and video link too, if supported). The two ends of the link will agree on audio codecs to use (how the audio will be transmitted), and each end tells the other end where to send the audio packets to. The audio travels on a different channel from the control channel (using Real Time Protocol - RTP), and your VoIP phone will send something like "send your audio to this IP address, I'm listening for RTP on (for example) ports 5008 - 5012". Each end agrees, and you have a successful phone call.
 
The section below on "Why is SIP a problem" explains how NAT breaks things.
 
==Why is NAT a problem with SIP?==
 
You have probably been directed to this page because you are having trouble with Voice over IP services (using SIP) when using a router or firewall which does some sort of Network Address Translation (NAT). This page tries to explain some of the issues, and why it is often a problem.
 
===SIP services we supply===
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