[SIPForum-discussion] Basic SIP Questions

[Fortunato Lacson]

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?

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.

Regards to all.


Fortunato
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