[SIPForum-discussion] Basic SIP Questions

[Vivek Batra]

Fortunato,
Comments Inline in RED.

--Vivek Batra

On Sun, Aug 17, 2008 at 9:58 PM, Fortunato Lacson <junlacson at gmail.com>wrote:

>
> Good day all. Like I mentioned before, I am very new in embracing SIP. My
> concern is the network portion of the project to ensure our bandwidth will
> be able to accommodate all the expected calls. I am doing rough calculations
> myself.
>
> My first question is so basic but I couldn't seem to find a straight answer
> to solidify what I understand from what I read. For a point to point SIP
> call to a SIP PABX, an RTP + RTCP session is established. This is calculated
> @ g.711 around 84Kbps on the WAN link. This RTP + RTCP session is only one
> way, and that another one will be established in the reverse direction. This
> means 2 X 84Kbps bandwidth out of my WAN link? So my 1 GigE WAN link can
> roughly accommodate (not considering other factors like tunneling, IPSec,
> etc); 1 Gig / (2 X84 Kbps) = simultaneous calls?
>

This is precisely true. However, you can increase the number of calls by
using codec's viz G.729 etc.

>
>
> Second question is, how can I monitor my SIP network in real time,
> specifically my DIDs, to ensure that they are not down on the Telco vendor
> end? Our business relies heavily on hundreds of DIDs, and in my current TDM
> setup, when a PRI is down I see it instantly.
>

How do you currently monitor the ISDN when it is down in your ISDN PBX?
How are you planning to interface SIP with ISDN? I mean, which gateway??

>
>
> Regards to all.
>
>
> Fortunato
>
>
>
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