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

[Graham Wooden]

On 10/30/08 5:49 AM, "Nuno Marques" <yangsengopenser at gmail.com> wrote:

> ...² i don't need to use mediaproxy in all calls, only in the NATed ones² ....
> 
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 don¹t 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.

With each g711u call leg, taking around 85kbps - that¹s 170 for each handled
call ... 85 in, 85 out ... you can really start eating away at bandwidth.

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
don¹t have to proxy the audio, the better.

And for the record, I have yet to come across a billing issue. I have
clients that pay for unlimited service and I have ones that pay on bundled
minutes plan.

Hope this helps and regards,

-graham
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.kamailio.org/pipermail/users/attachments/20081030/38ab992c/attachment.htm