PVDM4 max sessions & config

markbialik · ‎02-12-2018

Hello,

We have an ISR4331 with a PVDM4-128 on the motherboard. I'm new to the voice world, and have been tasked with finding out of the current PVDM4-128 is sufficient if we upgrade the total number of SIP paths we have with our provider. I inherited the configuration and am having a hard time understanding how to configure the 'max sessions' parameter if I need to.

Currently, we can have 90 concurrent SIP sessions with our provider. The DSP portion of our config looks like this:

!
voice-card 0/4
dsp services dspfarm
no watchdog
!

dspfarm profile 2 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 64
associate application SCCP
!
dspfarm profile 1 mtp
codec g729r8
maximum sessions software 160
associate application SCCP
!

When I try to change the value of 'max sessions' under profile 2, 64 is the highest I can select. That's my first question: why not 128? I've read about the different complexities and I'm wondering if it has to do with the order of the codecs specified? If we only configured G711 (or another low complexity codec), would it then allow me to choose 128?

We don't do any conferenceing in-house; we dial a 3rd party provider when we want to have conference calls (unless I'm completely misunderstanding what conferencing is all about).

If it matters, we do have CUCM, two AVG's for faxing, and a seperate VG connected to a PRI. The 4331 CUBE is our SIP router to the provider.

I'm having a hard time understanding how we could ever have more than 64 SIP calls with the 'max sessions' setting the way it is. A local resource told me that the PVDM4 is only used if transcoding needs to take place so we should be fine with the PVDM4-128 if we increase our SIP sessions from 90 to say, 110. I guess I'm having a hard time grasping this... I assumed some sort of transcoding is always going to take place, whether it's low, medium, or high complexity.

Sorry if these are stupid questions.... just trying to get my ahead around it all.

Thanks for any help.

R0g22 · ‎02-12-2018

In a CUBE setup, you don't need any DSP's. DSP are required for IP to Analog or TDM termination and/or vice-versa. A CUBE is an IP to IP gateway. So need for any DSP's to make your calls work. Your SIP calls/sessions will be solely dependent upon the router platform that you are using.

A transcoder is required if you have a codec mismatch. Let's say one IP leg is using G711ulaw while the second IP leg is going to use G729, the calls will not be able to complete until and unless a transcoder is invoked between the two legs. The transcoder will be invoked either by CUCM (SCCP) or can be invoked by CUBE itself (LTI).

So if you carefully design your network and make sure their are no codec mismatches then you would potentially not require any DSP/PVDM's on your router.

markbialik · ‎02-12-2018

Nipun,

Thank you for your reply. That's interesting. I know we're using the 4331 as a CUBE (I have the paper licenses that say CUBE for our 90 SIP sessions). I have no idea why the consultant who placed the order said we needed to upgrade the PVDM4 to a -128, then. Maybe this CUBE is doing more than a traditional CUBE? I will post a sanitized config below. Is there any transcoding between the router and CUCM? I also notice there is a config description for XCODE to UCCX. Thanks again.

!
voice service voip
ip address trusted list
ipv4 x.x.x.x 255.255.255.255
ipv4 x.x.x.x 255.255.255.255
address-hiding
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
rel1xx supported "rel100"
session refresh
header-passing
error-passthru
asserted-id pai
localhost dns:y.y.y.y
asymmetric payload full
early-offer forced
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
pass-thru subscribe-notify-events all
!
!
voice class uri FromCUCM sip
host x.x.x.x
host ipv4:y.y.y.y
host ipv4:z.z.z.z
!
voice class uri FromProvider sip
host a.a.a.a
host ipv4:b.b.b.b
voice class codec 100
codec preference 4 g711ulaw
codec preference 6 g729r8
!
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 3
!
!
voice class sip-profiles 101
request ANY sip-header Cisco-Guid remove
response ANY sip-header Cisco-Guid remove
request ANY sip-header User-Agent remove
response ANY sip-header User-Agent remove
request INVITE sip-header P-Asserted-Identity modify "(<.*:.*)(@.*)" "<sip:123456789@sip-provider.com>"
!
!
voice-card 0/4
dsp services dspfarm
no watchdog
!
!
mgcp
mgcp modem passthrough voip mode nse
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0/0
sccp ccm d.d.d.d identifier 11 version 7.0
sccp ccm e.e.e.e identifier 10 version 7.0
sccp
!
sccp ccm group 10
description XCODE for UCCX
associate ccm 11 priority 1
associate ccm 10 priority 2
associate profile 2 register 03-XCODE
associate profile 1 register 03-MTP
!
!
no ccm-manager fax protocol cisco
!
dspfarm profile 2 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 64
associate application SCCP
!
dspfarm profile 1 mtp
codec g729r8
maximum sessions software 160
associate application SCCP
!
dial-peer voice 90 voip
description Inbound SIP from Call Manager
session protocol sipv2
session target dns:our-call-manager-server-IP
incoming called-number .
incoming uri via FromCUCM
voice-class codec 100
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 91 voip
description Inbound SIP from SIP Provider
rtp payload-type cisco-codec-fax-ind 127
rtp payload-type nte 96
session protocol sipv2
session target sip-server
incoming called-number .
incoming uri via FromSIPprovider
voice-class codec 100
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 100 voip
description Outbound LD and Toll Free Calls to SIP Provider
destination-pattern 1[2-9]..[2-9]......
rtp payload-type cisco-codec-fax-ind 127
rtp payload-type nte 96
session protocol sipv2
session target sip-server
session transport udp
voice-class sip profiles 101
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 101 voip
description Outbound International Calls to SIP Provider
destination-pattern 011T
rtp payload-type cisco-codec-fax-ind 127
rtp payload-type nte 96
session protocol sipv2
session target sip-server
session transport udp
voice-class sip profiles 101
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 888 voip
description Outbound to CUCM
destination-pattern 888.......
session protocol sipv2
session target dns:our-cucm-server
session transport udp
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 800 voip
description Outbound to CUCM
destination-pattern 800.......
session protocol sipv2
session target dns:our-cucm-server
session transport udp
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
!
!
gateway
media-inactivity-criteria rtp
timer receive-rtp 86400
!

R0g22 · ‎02-12-2018

Your cucm pointing dial-peers are not explicitly configured for a codec, hence will default to g729. I am assuming this is a remote site so due to wan/bw consumption it would be that way.
Your UCCX must be doing G711ulaw. To compensate for the codec mismatch, you would need a transcoder.

R0g22 · ‎02-12-2018

To add to my post above, you can -

1. Check region settings on cucm for CTI pointing to UCCX. What is it set to ?
2. If you have access to UCCX, go to system > system parameter > codec.
See there what is it set to.

Neither cucm nor ccx can do transcoding natively. DSP resources on a IOS router will always be required for transcoding.

markbialik · ‎02-13-2018

Hi Nipun,

I checked the regions and device pools. All (but one) are set to the default:

Default

Use System Default (Factory Default low loss)

64 kbps (G.722, G.711)

384 kbps

2147483647 kbps

There is one entry:

G729 Everywhere

Use System Default (Factory Default low loss)

8 kbps (G.729)

None

2147483647 kbps

In UCCX, the information you requested is Codec parameter is set to G711. Thank you.

R0g22 · ‎02-13-2018

So that explains what I wrote above and your query for the need for a transcoder.

Chris Deren · ‎02-13-2018

You cannot design a SIP deployment for specific codec as most carriers will offer different codecs and for some calls single codec under specific conditions. For example ITSP1 customer A calls customerB (same ITSP) and customer A offers only G729 on outbound calls via EO, the ITSP will then relay only G729 to customerB as ITSPs will never do transcoding on their side (too expensive), thus customerB may have assumed they get all calls as G711 (ask for), but they are never guaranteed that and may require transcoders based on their region configuration and/or applications the calls go to (i.e. CCX defined with G711).

So, you ALWAYS need to account for some transcoding on ALL SIP deployments.

markbialik · ‎02-13-2018

Hi Chris,

Thanks for chiming-in. Every little piece of knowledge helps, and hopefully others reading this thread. I guess this answers the question of why the PVDM4 was specd-out to begin with. That makes me feel a lit better. So, I'm working through what Nipun said: In a CUBE setup, you don't need any DSP's" and what you said. I think, in fairness, Nipun did not have my full config when he answered the initial question. So, I am assuming with the CUCM, UCCX and SIP all in the mix, there definitely seems to be a need for transcoding. I'm still a little unsure about why the CUBE won't let me set a max sessions above 64 with a -128 PVDM4.

Chris Deren · ‎02-13-2018

The router does the calculation of how many sessions are available based on codec complexities, as different codecs require different amount of PVDMs. What codes do you have configured under the dspfarm?

markbialik · ‎02-13-2018

It's in one of the posts above, along with my theorizing that the max value had to do with defined codecs in the farm, but here it is again:

!
dspfarm profile 2 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 64
associate application SCCP
!
dspfarm profile 1 mtp
codec g729r8
maximum sessions software 160
associate application SCCP
!

I'm guessing by allowing the above codecs, the CUBE does an internal calculation of low/medium/high complexity codecs and decides on 64 as the best value? Thanks.

Chris Deren · ‎02-13-2018

remove following codecs from your transcoder profile as you dont need them:
codec g729ar8
codec g729abr8

markbialik · ‎02-13-2018

Thanks, Chris. Interesting. OK. I will try to get a downtime/change-window to do this today. Just for my own knowledge (and others who may be in the same boat), how do you know these are un-needed codecs? Are they legacy and not in use anymore? Do you think removing these may help bump that max sessions higher than 64? I'm not sure what increments these are typically given in. Thank you for your time.

tlewins · ‎04-01-2019

Hello Mark, I'm curious if you found out why these codecs weren't needed? Are they legacy and not in use anymore? Also, once you removed them, did it help to bump that max sessions higher than 64?