UC500 with SIP Trunks - can you transcode?

Unanswered Question
Sep 4th, 2008

I'm looking to deploy UC500 and use a SIP trunking provider for off-net inbound/outbound calls.

I want to use G.729 on the SIP trunk, however I know that CUE only support G711. Is it possible to configure UC500 to transcode the G.729 SIP Trunk calls to G.711 for CUE and use G.729 for all the phones? Does this work in the scenario where an inbound SIP call uses G.729 to a phone but then translates to G.711 if the call is sent to voice mail?

If so how much DSP resource is provided on the UC500 and how many transcoding sessions are supported. I'm not planning on using any BRI or FXO ports, although we may use a couple of FXS ports.

Your input would be appreciated.

Thanks,

Chris

I have this problem too.
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ccpagel Tue, 09/09/2008 - 07:24

This is the response I had from the partner helpline.

Thank you for contacting Partner Helpline Presales Support.

UC500 supports hardware configuration for 4 different user density: 8U, 16U, 32U and 48U. The 8U and 16U density models are shipped with PVDM2-32, whereas the 32U and 48U models are shipped with PVDM2-64. PVDM2-32 has 2 DSPs and PVDM2-64 has 4 DSPs. Without getting into too much technical details, let me just say that each DSP has 240 credits and g711 calls take 15 credit for each call, whereas g729 calls take 30 credits. Based on these credits, each DSP can support upto 16 g711 calls or upto 8 g729 calls or a mix of those.

It is important to understand what type of codec are being used in the deployment. This can vary if you have IP trunks (SIP and H.323). Also remember that the Cisco IP Phones have DSPs built in and can handle either g711 and g729 calls. So, if the SIP trunk or H.323 trunk is configured for g711 or g729, then for calls from these trunks to IP Phones, you won't need any PVDMs from UC500. However, one crucial consideration is Auto Attendant and Voicemail on CUE. The AA and VM can only handle g711 calls. If the SIP/H.323 trunk is configured for g729, then you will have to use some PVDM resources to create transcoding sessions (sessions that will convert g729 to g711).

If there are no IP trunks, then you will be using g711 to communicate w/ analog PSTN connections.

In either of the above two cases, additional DSP resources would be required depending upon the number of FXO/FXS/BRI ports. For example, if you have an 8-user model with 4 FXS and 4 FXO, it will use upto 8 g711 resources. With that you will be left with 8 more g711 resources plus an additional DSP. Similarly, if the expansion slot is used up with another 4 FXO ports, then you will be left with 4 g711 resources on the first DSP plus an additional DSP.

Hardware conferencing (Adhoc and Meet-me) requires 1 DSP that needs to be dedicated for conferencing. Transcoding sessions can be shared on the DSP that is being used for voice call.

In the SE Presentation, I've included slides that show how many transcoding/conferencing resources are available on a per DSP basis. Depending upon your deployment, you may have 1 DSP or 2 DSP or may be even 3 DSPs available for conferencing.

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