[Asterisk-Users] A solution for SIP and NAT

[Jim Flagg]

----- Original Message ----- 
From: "Michael Kane" <mkane at to-talk.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 02, 2003 7:37 AM
Subject: Re: [Asterisk-Users] A solution for SIP and NAT

<snip>

> That why I have looked into(implemented) such technologies
> like STUN and probably will be forced to purchase a SIP aware firewall that
> will spoof and re-arrange SIP messages destined for my proxy server.

<snip>

Correct me if I am wrong but I see a couple big disadvantages to this
solution.

1.  Voice latency can be significantly increased since all the RTP traffic has
to go through the VOIP providers NAT-proxy.  Even if you are calling your
next door neighbor, the traffic has to go all the way to the NAT-proxy and back.
Just ask one of the FWD NAT-proxy users in Europe what it does for sound
quality.

2.  The VOIP provider has to pay for all the bandwidth of the RTP steams rather
than just the small amount of traffic for call setup.