[Kamailio-Users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!

[Kristian Kielhofner]

On Thu, Oct 30, 2008 at 8:05 AM, Graham Wooden <graham at g-rock.net> wrote:
>
> I using 1.4.0 with the new NAT Transversal module, and it so far it handles
> all my NATed clients; even folks that have devices that don't support STUN
> (like the older Polycom IP Soundpoint phones). So in this case, the above
> statement is not true with me as I am not proxing their audio.
>
> I only proxy media under certain circumstances, like a court-ordered
> subpoena (CALEA),  call re-direction support (which I haven't got fully
> working yet), or virtual fax and other media services (voicemail, conf
> calls, etc) from which the audio goes straight to my asterisk machines.  And
> even with those, those are on a per-caller basis.

  While this is starting to get off-topic, I have to ask:

  Have you ever actually received a subpoena?  Are you a CLEC?  What
is your interconnection to the PSTN?

  The only reason I ask is because this sounds a little suspect.  In
most cases, telecoms CALEA is accomplished with LI capable software on
various media devices and a third party subscription based service
(like the one from Verisign) with direct or VPN access to twiddle the
SNMP bits to achieve compatibility with standards like
ATIS-1000678.2006.  You can't just trap RTP...  If you are an
"interconnected VoIP provider" you have to provide full CALEA
compliance to the relevant ATIS/TIA standards or figure out how you
can get someone to do it for you.  In many cases this can be easily
provided by the small handful of multi-billion dollar orgs that
provide these services in the US - Level(3), AT&T, Verizon Biz, XO,
etc.

  The only time I've ever been *aware* of a wiretap was when the
customer authorized the monitoring:  A couple of weeks ago a customer
of ours was hosting an event for a current US Presidential candidate
and the US Secret Service approached him asking for the contact
information of his provider (us).  The agent called me and faxed over
the authorization, which I verified and forwarded.  Other than that, I
never hear about it...

> With each g711u call leg, taking around 85kbps - that's 170 for each handled
> call ... 85 in, 85 out ... you can really start eating away at bandwidth.

  Isn't your bandwidth symmetric/full duplex?  How is 170kbps valid?

> Plus, I am finding that the call quality is a bit better when the audio goes
> directly from the NAT client straight to the PSTN provider. While we do
> operate our own network (AS / BGP, with two Tier1 and Tier2 providers), if I
> don't have to proxy the audio, the better.

  Totally makes sense in most cases:

- Depending on your connectivity
- Depending on your SIP/PSTN provider
- Depending on the customer's connectivity

..snip..

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com