Subject: [midcom] Comments regarding scalability issues with STUN/TURN/etc...

From: "Christopher A. Martin" <[email-masked]>
Date: Thu, 22 Apr 2004 11:55:25 -0500

Hi everyone, it has been awhile since I last posted.

These were some items posted to a common list for VoIP which I responded
to regarding considerations that should be made for any protocol being
used to address traversal of VoIP via middle boxes (in this case
firewalls).

Statements have been made that the solutions mentioned in the title
provide scalability over the emerging session border controller
technologies. 

Also the latest draft by Melinda brought up some additional points with
direct statements naming a couple of these vendors. I would like to
bring this discussion up a bit more to investigate the true scalability
factors.

I believe that we do need an IETF solution to the problem at hand that
does not present yet another issue to the mix, and that the current
protocol proposals need to consider scalability factors for a number of
reasons.

Let the attachments be a start to the discussion, while I formally write
up some more examples, which were between me, Johnathan, and Jiri.

Thanks in advance,

Chris

Subject: RE: [FWD] STUN server, when should I use it?
From: "Christopher A. Martin" <[email-masked]>
To: "'Free World Dialup - The Future of Dialing'" <[email-masked]>
Date: Thu, 8 Apr 2004 18:08:46 -0500

STUN servers use keep-alives between client and sip proxy to keep the
firewall/nat binding open for incoming media using null sip packets from
what I have seen.

Chris

-----Original Message-----
From: Free World Dialup - The Future of Dialing
[mailto:[email-masked]] On Behalf Of Jiri Kuthan
Sent: Thursday, April 08, 2004 1:38 PM
To: [email-masked]
Subject: Re: [FWD] STUN server, when should I use it?

At 07:47 PM 4/8/2004, Christopher A. Martin wrote:
>What about the keepalive issue? How does that scale?

I'm not sure to what keepalive issue are you reffering to,
but media relay scales imho much worse than keepalives.

Cheers,

-jiri

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Subject: RE: [FWD] STUN server, when should I use it?
From: "Christopher A. Martin" <[email-masked]>
To: "'Free World Dialup - The Future of Dialing'" <[email-masked]>
Date: Wed, 7 Apr 2004 23:35:29 -0500

STUN is necessary when there is no other means of traversing a NAT.
Without it no one would be able to send inbound calls your way, and
potentially no calls would work. 

There is a downside to STUN from the provider perspective in relation to
the keepalive mechanism used to keep the NAT/FW bindings fresh. This is
why session border controllers are used, such as the Jasomi outbound
proxy listed on the FWD website. Outbound proxies provide a lot more
scalability for the VoIP provider.

Chris

-----Original Message-----
From: Free World Dialup - The Future of Dialing
[mailto:[email-masked]] On Behalf Of boatman
Sent: Wednesday, April 07, 2004 9:24 PM
To: [email-masked]
Subject: Re: [FWD] STUN server, when should I use it?

Will I have less latency if I use the non-NAT configuration whenever
possible?  By non-NAT I mean setting the voice terminal as one would do
if it had it's own IP address and was not behind something, or behind
one of the SIP aware routers you mentioned.

When should one use the STUN settings?  Is STUN a work-around for being
behind a router?

At 04:59 PM 4/7/04, you wrote:
>"whenever you are behind a NAT router/firewall" is a bit too general.
Some
>NAT systems are "SIP aware" and poke around in the data stream
dynamically
>to change the embedded IP addresses.
>
>So if you have a SIP-aware NAT box/software, you can probably get away
>without doing anything "special".
>
>Don
>
>
>-----Original Message-----
>From: Free World Dialup - The Future of Dialing
>[mailto:[email-masked]]On Behalf Of boatman
>Sent: Wednesday, April 07, 2004 2:54 PM
>To: [email-masked]
>Subject: [FWD] STUN server, when should I use it?
>
>
>Referring to
>http://www.freeworlddialup.com/support/configuration_guide/configure_yo
ur_fw
>d_certified_phone/sipura_spa_2000
>
>I have a Sipura SPA-2000.
>
>I understand that I should use the Outbound Proxy settings whenever I
am
>behind a NAT firewall.  When should I or shouldn't I use the settings
for
>STUN?
>
>What is a STUN server?
>
>When I use the outbound proxy settings do my voice packets always go
through
>the proxy during conversation?  Wouldn't there be less latency if the
>packets were routed directly to the IP address of the other party?
>
>Is there a site where I can find the explanations of all this stuff?

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Subject: RE: [FWD] STUN server, when should I use it?
From: "Christopher A. Martin" <[email-masked]>
To: "'Free World Dialup - The Future of Dialing'" <[email-masked]>
Date: Wed, 7 Apr 2004 23:32:31 -0500

What about the issue of keepalives hitting the sip proxy, has that been
a problem? 

For a couple of users I wouldn't think there is an issue, but for
multiple users over say 100, there must be some degradation using the
stun server?

Chris

-----Original Message-----
From: Free World Dialup - The Future of Dialing
[mailto:[email-masked]] On Behalf Of Don
Sent: Wednesday, April 07, 2004 7:00 PM
To: [email-masked]
Subject: Re: [FWD] STUN server, when should I use it?

"whenever you are behind a NAT router/firewall" is a bit too general.
Some
NAT systems are "SIP aware" and poke around in the data stream
dynamically
to change the embedded IP addresses.

So if you have a SIP-aware NAT box/software, you can probably get away
without doing anything "special".

Don

-----Original Message-----
From: Free World Dialup - The Future of Dialing
[mailto:[email-masked]]On Behalf Of boatman
Sent: Wednesday, April 07, 2004 2:54 PM
To: [email-masked]
Subject: [FWD] STUN server, when should I use it?

Referring to
http://www.freeworlddialup.com/support/configuration_guide/configure_you
r_fw
d_certified_phone/sipura_spa_2000

I have a Sipura SPA-2000.

I understand that I should use the Outbound Proxy settings whenever I am
behind a NAT firewall.  When should I or shouldn't I use the settings
for
STUN?

What is a STUN server?

When I use the outbound proxy settings do my voice packets always go
through
the proxy during conversation?  Wouldn't there be less latency if the
packets were routed directly to the IP address of the other party?

Is there a site where I can find the explanations of all this stuff?

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