SIP, DSPs, transcoding, Call Manager question

Unanswered Question
Feb 5th, 2009

We have a new SIP trunk we are trying to implement in our Main Site that Verizon has seperated into two logical trunks.

Trunk 1 = 120 channels for customer IPCC IVR

trunk 2 = 200 channels for our company IPCC

All inbound SIP calls are g729.

Currently the existing system has the Main Site calls on all extensions as g711.

We have a 3845 Gateway with x4 64 DSP SIMMS installed.

When the system was designed, the idea was that only the 120 channels for would need to be transcoded from g729 to g711 for the IVR system because it is a voice banking system. It was thought the these calls needed to be g711 and not compressed locally once inside the building.

Now that we are testing, we are running out of DSP resources.

The Voice contractor is saying that they cannot get this to work this way.

They cannot get it to work with the Main Site extensions at g729, one of the symptoms is that we hear no ring back on out bound calls (everything else seems to work fine).

They are saying that all inbound and outbound calls int the Main site have to be transcoded to g711.

Should we be able to have call Manager handle the inbound SIP calls at g729 and do the IVR processing at g711?

Also, is there an appliance (if we need) that can provide DSP resources, rather than creating a DSP farm with a bunch of different routers?

I have this problem too.
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Nicholas Matthews Thu, 02/05/2009 - 15:58

The latest device that will add a great number of DSP resources is the VGD-1T3 box.

It supports cards with the AS5X-FC card, and you can put 3-4 cards with 6 PVMD2-64s in it. (A lot of DSPs).

Is the SIP trunk terminating on CUCM? A lot of customers find better luck using a CUBE between CUCM and their SIP provider. Then, SIP does the transcoding locally and you don't worry with CUCM based transcoding.

One possible reason you're not getting ringback is because you're requiring the use of Annunciator, which only uses G.711 I believe.



wilson_1234_2 Fri, 02/06/2009 - 16:35

Originally we had an edge router terminating the SIP trunk, then a CUBE (ip2ip gateway) router inbetween the edge router and call manager.

The voice guys had us remove the edge router and terminate the SIP trunk directly onto the CUBE router.

The CUBE is where the x4 64 PVDM SIMMs.

This is where the 128 channels should be available.

Nicholas Matthews Fri, 02/06/2009 - 17:04

If you take a look at the universal transcoding link, it shows 3845's can do up to 128 sessions.

If you have a 5400XM with the AS5X-FC cards, you can have up to 512 sessions on 12.4(20)T and later.

This may be something worth looking into if you want more sessions.



wilson_1234_2 Fri, 02/06/2009 - 18:04

The CUBE router is a 3845 with x4 64s for the 128 sessions I mentioned.

Are you talking about an additional 512 session + the 128 we have in the 3845?

Can you tell me a ballpark price of the 5400XM with the AS5X-FC cards?

Nicholas Matthews Fri, 02/06/2009 - 18:50

Disclaimer: TAC doesn't do pre-sales, so these numbers aren't anywhere near official, nor should they be used for reference.

Looking at the pricing it looks like the AS5400 is about 10-20% more than the 3845. You would need to buy 6 AF5X-PVDM2-64 cards, plus the AS5X-FC card. PVDM's are close to the same price as normal PVDM's, and the AS5X-FC card is going to be equivalent to be around the equivalent of adding another PVDM card.

I'm not sure what the differences are on the service contracts, because they vary quite a bit.

That would be an additional 512 on top of your existing 128 sessions, as it would be a separate box.




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