We have 3 sites. The head office is where all of our PRIs come into on our voice gateway (2951 ISR2). This router has 64 PVDM3 channels. So our branch sites use the wan for VoIP calls. Our branch sites have no PRIs or POTS installed locally, they use WAN for all voice traffic. Do I need PVDMs on those branch routers? Is there any benefit I could gain by having those resources available on the branch routers? I trying to figure out if I need the voice bundle with my routers or if i would be ok with a different option?
Unless you have TDM termination on the ISR they're not strictly needed.
You might have them just for HW CFB or XCODER as local resources if you need them, but that's pretty much it.
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In general, no voice interfaces = no PVDMs. So if you want a local PSTN connection then you would want PVDMs, and if you have FXS ports on the gateway for analogue stuff/faxes you need PVDMs.
I see lots of designs where people stick in PVDMs at the remote sites to 'transcode' but this isn't necessary unless you have stuff at the remote site that can't support the WAN codec. Typically you'll run G729 over the WAN, and phones, ATAs etc support that just fine. If you introduce a local server product (e.g. Unity Express) then you would need PVDMs to transcode G729-G711.
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While the other responders are strictly correct; if you have no voice interfaces you don't need DSP resources in a branch router and also that transcoding is rarely performed at the branch, there are other uses for the DSPs in the PVDMs. The one that I generally use is deploying a conference bridge at the remote site so that branch users do not have to go across the WAN to reach whatever central bridge you use.
The voice bundle also generally gets you SRST licenses which are usefull in building a survivable site allowing the telephones to register with the SRST router instead of turning into attractive bricks when the WAN is down. Unfortunately, an SRST site with no local PSTN access is not all that useful. How do you handle 911 calls? If a user at a branch dials 911 and the call goes out the central location you will have issues.
Finally, buying the voice bundle with some DSP resources is a good way to future-proof your deployment. SHould you ever need SRST or DSPs you will have them available without worrying about sending someone out to install the PVDMs and the Voice license. I believe it is generally cheaper to get the bundle than to buy it later.
What is HW CFB? Hardware Conference Bridge? I'm assuming you are referring to the conference button the phone. How does that use the PVDM resources, I never quite understood that?
Correct, we are talking about a hardware conference bridge. So say a branch phone created an ad-hoc conference with 2 other branch phones, without a local conference resource your Call Manager's conference bridge would be used. To create a hardware conference bridge on a branch router, you would configure SCCP (Skinny) which would allow the router to register it's resources with Call Manager. There's quite a few examples on Cisco's site for doing this. The bridge would then be assigned to the branch users by creating a MRG (Media Resource Group), putting it in a MRGL (Media Resource Group List) and then associating the MRGL with the device pool that the branch phones use.
Thanks, Gene Starr CCIE #8313
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What if instead of each branch site having a router capable of doing this we instead had a couple of high capacity routers located at our head office with high levels of PVDMs and they could register to call manager and be used for all of the MRG functionality?
That would be a potential solution, but would depend on your WAN capacity and what CODEC you run across the WAN. 3 people in a branch on a conference equals three calls across your WAN.
We would be using G.729 for the outgoing and incoming calls. Only time we would use G.711 is when going to CCX or Unity and Callmanager will handle the transcoding there.
Is it the region settings that determine what codec to use? Let's say I have region A and that region is the head office of our company. So i set that to g.711 and region B is a branch office. So in region A I set region B to be G.729. Now in region B I have region B setup for G.711 and region A setup for G.729. Does that make since?
Not quite - you don't apply a codec to a region, you assign a codec (or rather, maximum per-call bandwidth) to the relationship between two regions... so you pick Region A, and say 'from here to region B use G729 (8Kb max)'.
The default on CUCM is that 'intra region' calls goes G711/722, and inter-region calls go G729.
So is callmanager at that point handling XCODING? Or can I tell a voicegateway to do that and if so where in callmanager? I understand the regions now.
Call Manager will not transcode. Typically the devices that are talking will use the suggested CODEC. If you need to transcode (like if you a G729 call coming across the WAN that needs to go to UCCX, then you will need a hardware ttranscoder the is in whatever reqion that UCCX is in.
Yes, CODEC selection is governed by Regions (and regions are assigned by device pool membership). I assume you have a fairly recent version of Call Manager so when you create a region, say Region B, it is set by default to use G711 within that region and G729 with all other regions. You can change the default behavior by editing, but when you edit, you actually would set the CODEC that Region B would use with each other region. Typically the default is OK, but sometimes you would set a region to use G711 with all other regions (like UCCX) So using your example you would have something like this:
Region Region A CODEC Region B CODEC
Region A G711 G729
Region B G729 G711
To complicate matters, when you set the CODEC, what you are really setting is the Max bandwidth for a call, so for instance if you use G711, then G722 would also be allowed since it uses the same bandwidth.
How would I send traffic to the hardware transcoder or how would I verify that it is working correctly on the hardware transcoder. I can tell you in CUCM under media resources -> transcoder, I do have a transcoder setup but i'm not to sure about what it is doing or what traffic it is transcoding. It is registered and the transcoder type is "Cisco IOS Enhanced Media Termination Point". Sorry for all the questions, I have a system that has been dumped on me and my voice knowledge is not my best area ha. I'm learning as I go.
No problem Justin, glad to help if I can.
You can tell if a transcoder is being used by logging on to the router that hosts it and typing:
show sccp connections
which will show you all of the SCCP sessions currently running on the router. In your case if the regions are set up as you think, a call into UCCX from a branch phone should require a transcoder and the router command above should show you if it is happening. Another way is to use RTMT (Real Time Monitoring Tool) which is available in versions since 5.
What Call Manager version are you running?