Difference between revisions of "VoIP NAT"

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(Add a little more explanation)
(Start entering a SIP trace)
 
An incoming call works the same way. It's important to note that the VoIP server will send control messages to your local SIP port,
and your firewall connection tracker may have timed out the control channel if you haven't re-registered recently, or made an outgoing call recently - you either need to provide firewall rules to accept this control channel traffic, or send SIP Keep-Alives every (say) 120 seconds.
 
Here's an (edited) SIP trace of an outbound call being setup from 02083xxxxxx to 07973xxxxxx. The phone on public IP address 81.187.xx.xx sends:
 
INVITE sip:07973xxxxxx@voiceless.aa.net.uk;user=phone SIP/2.0
From: "aaisp" <sip:+442083xxxxxx@voiceless.aa.net.uk>;tag=xidd3faqpi
To: <sip:07973xxxxxx@voiceless.aa.net.uk;user=phone>
Content-Type: application/sdp
o=root 1510857313 1510857313 IN IP4 '''81.187.xx.xx'''
c=IN IP4 '''81.187.xx.xx'''
m=audio '''5010''' RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
 
The bold text shows the IP address and port number that the phone is listening on - telling the far end where to send audio.
 
The section below on "Why is SIP a problem" explains how NAT breaks things.