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

[Graham Wooden]

Quoting Kristian Kielhofner <kkielhofner at star2star.com>:


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

I have not directly received one, but know of folks that have. No, I  
am not a CLEC (nor plan to ever be one). My connection to the PSTN is  
SIP directly to the provider's SONUS switch.


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

Client -->[~85kbps] -->device proxying audio -->[~85kbps] -->PSTN

If both of those legs come in and out on your same Internet provider  
leg, well, that call is going to cost you 170kbps.  Since I do run BGP  
across multiple providers, I do have a fair bit of asymmetric routing,  
where the client may come in on my Tier2 and then sholve the call back  
out on my Tier1.  Still adds up to 170kbps no matter how you slice it.  
  But again, since I don't run full-time audio proxing, I don't have  
to worry about the bandwidth being absorbed like this anymore.

Why tack on another N amout of router hops and ms to the call if you  
don't need to?


>> 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

In which case, all my customers are under 120ms (all broadband or  
higher and not all are local), pretty much all under 12 hops to me and  
to the PSTN.  Works fine for me and I pay a fair bit of money to have  
my solid internet connections ;-)

Thanks,

-graham