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How much WAN bandwidth for call control?

Randall White
Level 3
Level 3

‎10-06-2004 02:43 PM - edited ‎03-13-2019 06:34 AM

Hi All,

I am looking at putting IP phones at some remote sites. These are connected to the Call Manager at the main site with Frame Relay (64kCIR/256K total). All calls at the remote sites will be to/from the local PSTN via T1 or analog lines on a local gateway. I will not be sending calls over the WAN.

How can I figure out how much bandwidth will be taken up by call control traffic? The remote sites will have about 10-12 phones, but call volume will be fairly high. How many simultaneous new calls can 64k support?

Thanks, Randy

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pbarman
Level 5
Level 5

‎10-06-2004 02:58 PM

As per Cisco recommendation you should use 5% of total voice payload bandwidth for signaling. Now the calls at the remote will definitely be g711, and 12 calls will consume about 900kbps, so signaling bandwith required for them would be ~45kbps, and that your FR link can easily accomodate.

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Randall White
Level 3
Level 3
In response to pbarman

‎10-07-2004 08:57 AM

Thanks,

Do you know what the "5% rule" is based on? Obviously the signaling traffic will be essentially zero once the call is established. My main concern is a large number of calls coming in at the same time. I suspect I'll have to test this out in a lab situation.

Randy

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MICHAEL BASTINCK
Level 1
Level 1

‎10-07-2004 06:06 AM

Hi Randy

We have some sort of same situation over here. We have some remote sites and I do not want that any call will be send over the WAN. Every site has it's own gateway but I have a problem with my extension mobility users. If they login and want to make a phone call, the call will be send over the WAN to ther original user's gateway. Do you have any idea how this user can make an outbound call via the gateway at the location he logged in?

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allan.wells
Level 3
Level 3
In response to MICHAEL BASTINCK

‎10-08-2004 12:23 AM

Thats an issue with Extension mobility its only good in a campuss enviroment.

If you have a centralised cluster and spokes connected via wan links with EM then your going to have issues if the EM uses roam to remote sites.

As you know the roaming users would be decrementing the location BW of the site they were configured in.While the actual RTP stream is eating away the LLQ elswhere.

If you provisiioned your LLQ way over spec then it might be alright.But very unlikely in most countries can you afford to do this.

Ive heard that CM 5 on linux is going to fix this issue lets wait and see.

The other option and better one would be to have sip proxies that communicate with each other and allow users to roam free.

Still I dont know how you would dimension your LLQ in this design with 100's or 1000's of users roaming.

Perhaps there is a smart CAC mechanism associted with SIP proxy worth investigating.

As SIP is really starting to fly now !

The other easy option and the one I use and recomend is that Users use ccmuser to FWD there phones to the desk there at this way CAC and LLQ are not effected.

CoS on Phones isnt an issue if you use FAC so all phones cann dial all numbers but you need a FAC

Its a work around

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