Are transcoding resources necessary on this case?

[filipeleite]

Hey Guys,

I have an E1 r2 digital configured with only 10 channels.

Although i have 30 channels configured on the PSTN side i could only configure 10 channels due DSP resource not being avaliable for transcoding and conference at the same time.

So, my question is, is it possible to complete remove transcoding and conference to make avaliable the full 30 channels?

I know the drawbacks of not having conference but the transcoding resources are what i'm afraid of.

Since the call will arrive using PCM from the provider and on the LAN i'm using G711 i think that we will be good.

What do you think?

Thanks in advance.

Right now i have only a single PVDM as follow:

NAME: "PVDM3 DSP DIMM with 32 Channels on Slot 0 SubSlot 4", DESCR: "PVDM3 DSP DIMM with 32 Channels"

PID: PVDM3-32          , VID: V01 , SN: FOC144651K1

[islam.kamal]

Hi

Transcoding compresses and decompresses voice streams to match endpoint-device capabilities.

Transcoding is required when an incoming voice stream is digitized and compressed (by means of a

codec) to save bandwidth, and the local device does not support that type of compression

When do you need Transcoding?

• Ad hoc conferencing—One or more remote conferencing parties uses G.729.

• Call transfer and forward—One leg of a Voice over IP (VoIP)-to-VoIP hairpin call uses G.711 and the

other leg uses G.729. A hairpin call is an incoming call that is transferred or forwarded over the same

interface from which it arrived.

• Cisco Unity Express—An H.323 or SIP call using G.729 is forwarded to Cisco Unity Express.

• Cisco Unity Express supports only G.711, so G.729 must be transcoded.

• Music on hold (MOH)—The phone receiving MOH is part of a system that uses G.729. The G.711 MOH

is transcoded into G.729 resulting in a poorer quality sound due to the lower compression of G.729

I think you will not use it ,so go ahead .

Thank you

please rate if this will help

[filipeleite]

Thanks!

[Ayodeji Okanlawon]

It all depends on what your setup is...

IP---CUCM----h323--Gateway----PSTN

If in this set up, the region between your phone and gateway is set to G711 and your inbound dial-peer to the gateway from CUCM is set to G729, You will need a xcoder.  However if you use a voice class codec on your voip dial-peer then you wont need a transcoder....

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[filipeleite]

Thanks!