Thanks for reply. Sorry, I didn't get quite well your suggestion. Without a doubt the QoS will be configured on both sites and in fact we will have one ADSL link on both sides dedicated to voice traffic. I am thinking if we had T1 at the main site with CUCMBE and ADSL on remote site where we install SRST, we would still be at mercy of ISP when calling from the remote site to the main site, we alway are. When your traffic passes vearious ISP, there is no QoS guaranteed. Shame we cannot get T1 on the main site so we are at mercy of ADSL.

Apart from the codec calculations I will aslo need to know the vpn overhead, call signaling bandwidth, voice masseges pulled from Cisco Unity on CUCMBE etc etc. I know we have only 768kbps uplink on the main site for all the voice traffic from the remote site but is that going to be enough?

If they play voicemails through the phone, its still a voice call. If they are downloading from Exchange its still considered data. Make sense?

At G.729, your voice traffic per call is still 40k (roughly) per call. That is up/down traffic. You have plenty of bandwidth as long as your data is lower priority between the sites. If you do not, as soon as someone drags a big file, all voice traffic will halt or become garbled.

As for overhead VPN traffic, depends on your encryption. Is not that much, but here is a quick google search and table:

http://tisc-insight.com/newsletters/39.html

You are dealing with BITs not KBs with overhead.

With any voice, you need to calculate overhead anyways, but also a little bit more room for worst case. Your worst case is 20 calls, but highly doubt everyone will be on the phone with the central site.

Remember, your central site still needs to have bandwidth to supply voice to remote site.

Makes sense, thanks. I am thinking what solution would be best here, either CME or SRST. I will have around 20 analog phones conecting to either VG224 or another SIP voice gateway with FXS ports. Would you be kind and suggest what would work best in this scenario? In remote location we have 3 analog phone lines so our CME or SRST will be running on 2901 with PVDM2-16 and VIC2-4FXO. In this scenario would we be better off with CME or SRST configuration? With SCCP or SIP protocol? The phones will run basic funcions, call pickup, transfer, park and no hunt groups. Thanks in advance.

All depends on cost. VG224 will give you the most analog devices for the price. Adding FXS cards to a 29xx will only allow a certain number, I think its 16 analogs, cant recall.

So if you only need less than 16, Id say a 29xx router is fine.

VG224 will also work, it can handle SRST also, running SCCP/MGCP, etc.

It really does not matter if its SIP or SCCP. It still is going to use bandwidth. For the gateway, I would not use MGCP, and use H323. MGCP may "bounce" a lot with that type of connection.

So VG224 would work just fine without ISR 2901 running SRST? I didn't know that VG224 can work as SRST. How can it acomodate FXO connections in it? Can I install a VIC card in it? I don't think so... So if I could I would just go for one VG224 and some SOHO router to do QoS and VPN don't you think? Thanks.

IF you need FXO, then you need a 29xx router with FXO cards.

And yes, before the traffic hits the DSL line, you have to give priority to voice, and drop the data