CUCMBE and SRST over ADSL 4Mbps/768kbps on both ends

Unanswered Question
Apr 26th, 2010

Has anybody set up a CUCME or CUCMBE over ADSL link? What I am planning  on implementing is a CUCMBE in the main site with SRST on the remote  site. In remote site I will have around 20-25 phones. Maximum amout of  calls between the site at pick will be around 5. The sites will be conected over IPSec tunnel, possibly GRE. Will I have enough  bandwidth for call signaling and the voice traffic? I was also thinking  to pass all incoming external calls in the remote site to AA on CUCMBE  in the main site, how would that work? How would the ADSL perform here?  Thanks.

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Tommer Catlin Mon, 04/26/2010 - 08:14

You can do it, but calls between the sites, you cant expect any QOS with DSL.  Once it hits the DSL provider network, there is QOS.. You can prioritize the traffic at each end when it enters and leaves the site (voice over data) but again.. you are at the mercy of the provider.


Your rough numbers:


20 simultaneous calls between sites @g.729 codec, each call is around 40k (roughly).


40k x 20 calls = 800k.  (not even a 1meg) 



So even at 5 calls, you are fine.  Its just the QOS part.  You never know because its not guaranteed. 

remi-reszka Mon, 04/26/2010 - 08:51

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?

Tommer Catlin Mon, 04/26/2010 - 09:03

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.

remi-reszka Mon, 04/26/2010 - 11:44

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.

Tommer Catlin Mon, 04/26/2010 - 11:51

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.

remi-reszka Mon, 04/26/2010 - 12:00

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.

Tommer Catlin Mon, 04/26/2010 - 12:09

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

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