DSP PVDM2_64 on 2821 router

Answered Question
May 28th, 2008
User Badges:

Hi Networkers,


im facing an issue with my 2821 router DSP configuration. am trying to configure DSP , i have built in PVDM2-32 on voice card 0 , and installed PVDM2-64 on PVDM slot 1. im not able to configure the voice-card 1 i got the following error:

ft-worth(config)#voice-card 1

^

% Invalid input detected at '^' marker.


ft-worth(config)#voice-card ?

<0-2> Voice interface slot #


ft-worth(config)#voice-card


while the diag command showed that the PVDM-64 in PVDM slot 1 detected and installed see blow

VDM Slot 0:

32-channel (G.711) Voice/Fax PVDMII DSP SIMM PVDM daughter card

Hardware Revision : 4.0

Part Number : 73-8539-05

Board Revision : A0

Deviation Number : 0

Fab Version : 04

PCB Serial Number : FOC110254DQ

RMA Test History : 00

RMA Number : 0-0-0-0

RMA History : 00

--More-- Fab Version : 04

PCB Serial Number : FOC110254DQ

RMA Test History : 00

RMA Number : 0-0-0-0

RMA History : 00

Processor type : 00

Product (FRU) Number : PVDM2-32

Version Identifier : V01

EEPROM format version 4

EEPROM contents (hex):

0x00: 04 FF 40 03 EE 41 04 00 82 49 21 5B 05 42 41 30

0x10: 88 00 00 00 00 02 04 C1 8B 46 4F 43 31 31 30 32

0x20: 35 34 44 51 03 00 81 00 00 00 00 04 00 09 00 CB

0x30: 88 50 56 44 4D 32 2D 33 32 89 56 30 31 20 D9 02

0x40: 40 C1 FF FF FF FF FF FF FF FF FF FF FF FF FF FF

0x50: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF

0x60: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF

0x70: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF


PVDM Slot 1:

64-channel (G.711) Voice/Fax PVDMII DSP SIMM PVDM daughter card

Hardware Revision : 4.0

Part Number : 73-8541-05

Board Revision : B0

Deviation Number : 0

Fab Version : 04

PCB Serial Number : FOC1120339H

RMA Test History : 00

--More-- Board Revision : B0

Deviation Number : 0

Fab Version : 04

PCB Serial Number : FOC1120339H

RMA Test History : 00

RMA Number : 0-0-0-0

RMA History : 00

Processor type : 00

Product (FRU) Number : PVDM2-64

Version Identifier : V01

EEPROM format version 4

EEPROM contents (hex):

0x00: 04 FF 40 03 EC 41 04 00 82 49 21 5D 05 42 42 30

0x10: 88 00 00 00 00 02 04 C1 8B 46 4F 43 31 31 32 30

0x20: 33 33 39 48 03 00 81 00 00 00 00 04 00 09 00 CB

0x30: 88 50 56 44 4D 32 2D 36 34 89 56 30 31 20 D9 02

0x40: 40 C1 FF FF FF FF FF FF FF FF FF FF FF FF FF FF

0x50: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF

0x60: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF

0x70: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF


WIC Slot 0:

VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1

Hardware Revision : 0.0



any idea

Correct Answer by Chris Deren about 9 years 1 month ago

It is considrered a transcoding, however Cisco phones (not sure about Linksys) can do the transcoding natively. You may however require MTP resources for DTMF processing accross SIP trunk depending on your carrier. This is something you should ask the carrier about.


HTH, plese rate all useful posts!


Chris

  • 1
  • 2
  • 3
  • 4
  • 5
Overall Rating: 5 (1 ratings)
Loading.
Chris Deren Wed, 05/28/2008 - 08:05
User Badges:
  • Super Silver, 17500 points or more
  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

voice-card 1 is actually Network Module (NM) slot, all PVDMs slots in the chassis belong to voice-card 0.


HTH,


Chris

t-alkhalidi Wed, 05/28/2008 - 09:02
User Badges:

ops thank you so much. the problem when i do issue command show dspfarm dsp all i see 0 dsp available. by defualt the 2821 has 3 PVDMS slots on the mother board. do i need to configure the router and PVDM for my E1 30 channels and 4FXO ports? or resources of DSP will be allocated automatically without configuration?

Chris Deren Wed, 05/28/2008 - 09:07
User Badges:
  • Super Silver, 17500 points or more
  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

The DSPs will be allocated dynamically, all you need to do is configure the controller for E1 and the IOS will check if there is sufficient number of DSPs, if not it will give you an error. You can use the remainder for Conferencing and/or transocding if desired.


HTH,


Chris


Chr

t-alkhalidi Wed, 05/28/2008 - 09:19
User Badges:

thank you chris so much. your info is really helpfull. you know i wonder if we need transcode or MTP for 3rd party linksys SIP phones to convert the G.711 MU which all phones use to G.711 A Law which my telco provider use? from G.711 MU to G.711 Alaw does it considered as transcode session?

Correct Answer
Chris Deren Wed, 05/28/2008 - 09:47
User Badges:
  • Super Silver, 17500 points or more
  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

It is considrered a transcoding, however Cisco phones (not sure about Linksys) can do the transcoding natively. You may however require MTP resources for DTMF processing accross SIP trunk depending on your carrier. This is something you should ask the carrier about.


HTH, plese rate all useful posts!


Chris

t-alkhalidi Wed, 05/28/2008 - 10:01
User Badges:

thats great:) Cisco phones transcode from G.711 Mu to G.711 A which E1 use localy. i mean its free my DSP so i can use the DSP for other conf and MTP

Chris Deren Wed, 05/28/2008 - 10:12
User Badges:
  • Super Silver, 17500 points or more
  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

Let me clarify, the phone will transcode natively between themselves and other supported devices, if you need to make a call across E1 to PSTN, you will still need PVDMs to support the TDM channels on the E1. So, if you run the GW in flex-mode for DSPs (default) and the call arrives at the GW and need to be sent accross WAN to a location that is using G729 (based on regions in CM), the phone will negoitiate G729 codec with the GW, no additional transcoders will be required on top of the DSPs allocated to the E1 channels.


Chris

craig.pollitt Wed, 05/28/2008 - 12:14
User Badges:

cderen,


I have a 2821 w/ two E1 ports and two PVDM2-64 with 4 DSPs each. You mention that the remaining DSP resources can be used for conferencing and such. How exactly is this achieved?


Thanks,

Craig L. Pollitt

t-alkhalidi Wed, 05/28/2008 - 12:26
User Badges:


check this link to configure transcoding and conferencing. it has sample example and all what you need:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008084fe1f.shtml


i know that i need MTP if i have SIP trunk ,

and for 2 site compressed callE.g G711 to G.729 you need DSP at both sides routers even if the phones are Cisco phones.


Chris Deren Wed, 05/28/2008 - 13:00
User Badges:
  • Super Silver, 17500 points or more
  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

You do not need transcoders between IP phones, as I said the phones will negotiate codec based on the region configuration within CM. You would need transcoders for example if you have CRS installed (CRS can only be installed with one codec), or if you decided not to allow Unity to use G729 codec and do native transcoding on the server. Phones themselves have built in DSPs that allow them to negotiate any codec.


Chris

t-alkhalidi Thu, 05/29/2008 - 08:08
User Badges:

thats great, do i need to enable any configuration to enable the DSP or direct IP or hairpining, or its there by defualt.

craigt Fri, 06/13/2008 - 16:07
User Badges:

cderen,


Say I have site 'A' with 2821 where a Pub, Sub and Unity (doing G.711) reside and site 'B' is a branch with another GW terminating a PRI. My understanding


is when one enables a T1 controller for a PRI, it dynamically allocates channels into dsp resources in flex mode. The channels can then take the TDM end


from the PSTN and convert it into which ever codec is required to match opposing endpoints(voice-class codec/region considered). So, if a call comes into


site B on the PRI and it converts from TDM to codec G.729 to go over the WAN to a site 'A' endpoint, it does so without the need for additional DSP resources


from the site 'B' GW (transcoder) thus only using the allocated DSP channel resource. At site 'A', depending on the endpoint (Unity), it may require


transcoding back from G.729 to G.711.


So, if one is doing a calculation in the DSP Calculator and it is a single centralized site deployment doing G.711, one can enter all calls for each channel


at G.711 and will need the fewest DSP resources. However, if one has a multi-site centralized deployment and uses G.729 between sites/regions, one has to


assume several or perhaps all calls could be sent from site 'B' to Site 'A' and the DSP calculator will indicate more DSP resources are required. However,


one cannot base a DSP calculation for Site 'B' at G.711 per call and assume by adding transcoding resources to the DSP calculator that the transcoders will convert the TDM to G.711 DSP channel to G.729 to traverse the WAN to site 'A' endpoint.


Does this sound plausible/correct/right?


Regards,


Craig

Chris Deren Sat, 06/14/2008 - 12:45
User Badges:
  • Super Silver, 17500 points or more
  • Hall of Fame,
  • Cisco Designated VIP,

    2017 IP Telephony, Contact Center, Unified Communications

You are right on the target. To your last paragraph you won't need transcoders as the GW will natively use G729 when running in flex mode, the DSP calulcator allows you to sepcify how many G729 vs G711 you will use on the VWIC card.

You are also correct in regards to Unity, however Unity can perform native software transocding by changing a paramter, using hardware transcoders would be a better option as native software transcoding puts stress on the server CPU.


Chris

Actions

This Discussion