[Asterisk-Users] A solution for SIP and NAT

Michael Kane mkane at to-talk.com
Wed Jul 2 14:04:47 MST 2003


Hey Jim , you are correct in respect to the Service provider must pay for
the bandwidth as I/we will be hair pinning calls back into the Internet.  As
far as voice quality is concerned (which is my biggest concern) the
solution(box) FWD uses will not be the solution I will implement.  I am
seriously looking/talking with another vendor.  But, back to the point,
unless a SP has a subscriber base that is technically savy we/I really have
no other choice.  Again I have a STUN server implemented, but, SNOM and
Grandstream are the only hardphone vendors that ship their products with a
stun client(that I know of).  Beleive me, I wish this wasn't something I had
to think about.

regards Mike



Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
www.to-talk.com
508-295-2826
----- Original Message ----- 
From: "Jim Flagg" <flaggj at comcast.net>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 02, 2003 4:17 PM
Subject: Re: [Asterisk-Users] A solution for SIP and NAT


> ----- 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.
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> Asterisk-Users at lists.digium.com
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>




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