DSP MTP Config

Answered Question
Aug 18th, 2009

VGateway-3825#sh dsp all

Dspfarm Profile Configuration

Profile ID = 1, Service = CONFERENCING, Resource ID = 1

Profile Description :

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP Status : ASSOCIATED

Resource Provider : FLEX_DSPRM Status : UP

Number of Resource Configured : 8

Number of Resource Available : 8

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required

Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required

Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required

Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required

Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required

Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required

SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0 5 1.1.1 UP N/A FREE conf 1 - - -

0 5 1.1.1 UP N/A FREE conf 1 - - -

0 6 1.1.1 UP N/A FREE conf 1 - - -

0 6 1.1.1 UP N/A FREE conf 1 - - -

0 7 1.1.1 UP N/A FREE conf 1 - - -

0 7 1.1.1 UP N/A FREE conf 1 - - -

0 8 1.1.1 UP N/A FREE conf 1 - - -

0 8 1.1.1 UP N/A FREE conf 1 - - -

Total number of DSPFARM DSP channel(s) 8

VGateway-3825#

As per my understanding we must have a MTP Resource whenever we terminate a PRI or any other PSTN on a Voice Gateway.

However I recently encoutered a setup where No Dsps are used for MPT purposes and the DSPs are for Conf. only. Yet its working.

Did I miss something here or its all the calls are running on the G711 codecs, btw same is true on all other remote sites as well (I checked CCM to find no MPT resources than the Software MTP)

Is there a way to check what Codecs are being used for calls across the WAN Links?

Any help would be greatly appreciated.

Regards

Wilson Samuel

I have this problem too.
0 votes
Correct Answer by Nicholas Matthews about 7 years 3 months ago

Hi Wilson,

Yes I can confirm this.

If you have cards like VIC2-2FXO the actual voice ports will not even show up on the router unless you have DSPs. I recently had them change it so that this prints out an error message, so that may be more clear in the future.

For T1s - the router will spit back a message saying that you do not have enough DSP resources to configure X channels if you do not have DSPs. DSP allocation is best viewed with the 'show voice dsp group all' command.

-nick

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Overall Rating: 4.3 (5 ratings)
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Jaime Valencia Tue, 08/18/2009 - 08:14

Your understanding is wrong, read this to understand MTPs

Media Termination Point (MTP)

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/media.html#wp1046314

You should already be using DSPs for Voice Termination, that's why the PRIs work without any MTP profiles.

More info here:

Configuring Enhanced Conferencing and Transcoding for Voice Gateway Routers

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html

HTH

java

if this helps, please rate

Wilson Samuel Tue, 08/18/2009 - 08:18

Hi Java,

Thanks for your quick reply.

Agreed MTP is not a requirement for PRI/PSTN termination and its for H323 implementation.

What about Voice Termination (I guess I got confused with it) also I recall clearly when the 2VWIC-T1 card failed to initialize when at one site we didnt have enough DSP.

Edit:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/media.html#wp1045532

Regards

Wilson Samuel

Wilson Samuel Tue, 08/18/2009 - 10:13

I have no idea! I'm a newbie to UC/IPT.

Please appreciate my limited knowledge in UC/IPT.

I'm still puzzled by this DSP configuration on one of the rotuers.

Anyone cares to explain? Please?

Regards

Wilson SAmuel

Wilson Samuel Thu, 08/20/2009 - 07:17

No answers yet?

Surprising! Did I ask any insignificant question here?

Regards

Nicholas Matthews Thu, 08/20/2009 - 09:10

Hi,

Your problem description isn't very clear here, other than you don't understand DSPs very well.

A few things that may help:

DSPs are used to terminate an analog connection into RTP packets. This means all PRI/FXO/FXS/CAS/etc use DSPs. When they are doing termination, they will use any codec necessary albeit at different complexities.

DSPs are also used in transcoding to turn one codec into another.

They are also used in conferencing to multiplex and balance multiple streams into a single stream.

For MTPs, it is slightly different. The MTP is simply a place where RTP packets come in, and they go back out. This can be done in either software or hardware, because there are no complex algorithms or requirement for hardware processing to pass a packet in that we have already received. All the other scenarios are centered around changing audio-level parameters and codecs.

The reason you use MTPs is so that you can negotiate media (audio) at a early stage in signaling. When a phone creates a phone call, it doesn't immediately tell CUCM where it would like to receive audio (ip/port). When CUCM talks to devices that want to know where to send audio as soon as the call is sent, it has to pick a 'mediator' of sorts which turns out to be the MTP. This is the reason it's required for fast-start.

There are also certain DTMF methods that require for CUCM to be able to talk to the device that is receiving the media to learn things about what's going on in the media stream. If the MTP was not involved it would not be able to know about them.

Additionally, it can be used in transfer and hold scenarios as more of an 'anchor' for the call, to make sure CUCM always knows what IP/port to negotiate media with.

Hope this helps.

-nick

Wilson Samuel Thu, 08/20/2009 - 10:37

Hi Nick,

Thanks for your reply and my apologies for not writing it clearly. Let me re-attempt it:

All I want to know is about Voice Termination (I was incorrectly referring MTP, where in I must have used Voice Termination, somehow got jumbled in)

Now, if I have to terminate a PRI on a Gateway, definitly I will need the DSPs for the Voice Termination, Correct?

If yes, how do I configure it?

Because in my case, though the PRIs are terminating on the Gateways, I dont see any configuration for the Voice Termination function of the DSPs (its for Conferencing only).

Hope I have expressed myself clearly over here.

VGateway-3825#sh dsp all

Dspfarm Profile Configuration

Profile ID = 1, Service = CONFERENCING, Resource ID = 1

Profile Description :

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP Status : ASSOCIATED

Resource Provider : FLEX_DSPRM Status : UP

Number of Resource Configured : 8

Number of Resource Available : 8

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required

Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required

Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required

Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required

Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required

Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required

SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0 5 1.1.1 UP N/A FREE conf 1 - - -

0 5 1.1.1 UP N/A FREE conf 1 - - -

0 6 1.1.1 UP N/A FREE conf 1 - - -

0 6 1.1.1 UP N/A FREE conf 1 - - -

0 7 1.1.1 UP N/A FREE conf 1 - - -

0 7 1.1.1 UP N/A FREE conf 1 - - -

0 8 1.1.1 UP N/A FREE conf 1 - - -

0 8 1.1.1 UP N/A FREE conf 1 - - -

Total number of DSPFARM DSP channel(s) 8

VGateway-3825#

Nicholas Matthews Thu, 08/20/2009 - 16:20

"Voice termination for conferencing only" doesn't make any sense.

In order to conference you have to terminate voice first and foremost. That's the primary feature, and the configuration is just dial peers and other gateway configuration. There is no explicit DSP configuration, just the requirement of having DSPs.

Conferencing is a supplementary feature. You will need more DSPs for conferencing also.

If you simply want to conference ONLY IP phones then you can do this with the software conference bridge on CUCM using G711.

Check out the DSP calculator, maybe it can offer more information.

http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl

hth,

nick

Wilson Samuel Fri, 08/21/2009 - 03:59

"There is no explicit configuration, just the requirement of having DSPs"

I believe this is the crux point.

I thought just like MTP, Conf Bridge or Transcoding one does need the DSPs to be configured before it can be used.

Now I understand, one MUST have a physical presence of the DSPs (PVDM2-32 et al) on the Router, in order to have the PRI to be terminated.

Once again, Thanks for all your efforts, this I'm marking +5.0, and if you could please confirm it, I will be obliged to give +5 (Author's issue resolved)

Thanks a Lot!!!

Regards

Wilson Samuel

Correct Answer
Nicholas Matthews Fri, 08/21/2009 - 06:30

Hi Wilson,

Yes I can confirm this.

If you have cards like VIC2-2FXO the actual voice ports will not even show up on the router unless you have DSPs. I recently had them change it so that this prints out an error message, so that may be more clear in the future.

For T1s - the router will spit back a message saying that you do not have enough DSP resources to configure X channels if you do not have DSPs. DSP allocation is best viewed with the 'show voice dsp group all' command.

-nick

Wilson Samuel Fri, 08/21/2009 - 08:32

Fantastic!! Thank you Sir!

I have had struggled with this for a while, actually no book/document explictly mentions that one Does NOT needt to configure for Voice Termination Process (ofcourse they just move on to configure it for MTP or Codec or Conf. reasons)

Regards

Sam Wilson

balasundaram.k_2 Fri, 08/21/2009 - 01:36

Hi Nick,

Many Thanks for the explanation on the MTPs. The best description of the functionality of MTP, i have read so far!!!

+5 for you!!

Cheers,

Bala.

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