[asterisk-users] NAT traversal packet loss measurement

[Matt Riddell]

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Yitzhak Bar Geva wrote:
> How can one measure the effect of NAT traversal packet loss?
> We currently have no solution for NAT traversal for our SIP clients. There
> is no doubt that packets are getting lost. What is not clear is how much
> damage this does. On the face of it, everything seems fine. Could this be
> so? Perhaps we're suffering a degradation in quality or our call setup times
> could be improved. How can we measure this?
> What's the simplest method of preventing packet loss due to NAT traversal in
> a SIP environment?

NAT is unlikely to cause a percentage of packets to get lost.

Normally you'd have one way audio if NAT was causing a problem (i.e.
100% packet loss).

The only other situation in which it might happen is where the NAT
router decides to close a port mapping (thereby blocking incoming calls
to the customer's device).

But if you're looking for packet loss there are a number of other things
to check first.

I wouldn't do VoIP across the WAN without at least some packet shaping
but hey.

- --
Kind Regards,

Matt Riddell
Director
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