Is PVDM require on Cisco Router ISR4331

[abdulwadood]

I have Cisco Router ISR4331 with UC IOS, A-Flex License License.

For Voice, i got the connection from ISP i.e. SIP TRUNK line and he provided these codecs , g711 ,g729/t.38 port 5060

Please find the below config:

voice service voip
ip address trusted list
ipv4 10.180.12.114
ipv4 172.16.0.0 255.255.255.0
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
trace
sip
header-passing
error-passthru
registrar server
early-offer forced
midcall-signaling passthru
call-route history-info
sip-profiles 10 inbound
!
voice class codec 1
codec preference 1 g711alaw
!
!
voice class sip-profiles 10
request INVITE sip-header From modify "10.180.12.11" "10.180.12.112"
response ANY sip-header Contact modify "10.180.12.11" "10.180.12.112"
!
!
!
!
!
voice register global
mode cme
source-address 172.16.53.2 port 5060
max-dn 20
max-pool 20
load 8851 sip88xx.14-2-1-0101-26
authenticate register
authenticate realm ims.mobily.com.sa
timezone 31
url directory url
tftp-path flash:
create profile sync 0052552455220496
auto-register
!
!
voice register dn 1
number 7771
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch

label 966115127771
mwi
!
voice register dn 2
number 7772
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch

label 966115127772
mwi
!
voice register dn 3
number 7773
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
name TA
label
mwi
!
voice register dn 4
number 7774
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
label 966115127774
mwi
!
voice register dn 5
number 7767
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
name Iyas Alhabib
label 966115127767
mwi
!
voice register dn 6
number 7766
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
name Security Operation
label 966115127766
mwi
!
voice register dn 7
number 7768
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
label 966115127768
mwi
!
voice register dn 8
number 7765
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
name Wado
label 966115127765
mwi
!
voice register dn 9
number 7769
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
name Khaled
label 966115127769
mwi
!
voice register dn 10
number 7770
call-forward b2bua busy 100
call-forward b2bua noan 100 timeout 20
allow watch
name Abdul
label 966115127770
mwi
!
voice register pool 1
busy-trigger-per-button 2
id mac E089.9DFB.CE42
type 8851
number 1 dn 1
dtmf-relay sip-notify

!
voice register pool 2
busy-trigger-per-button 2
id mac 0027.902A.2972
type 8851
number 1 dn 2
dtmf-relay sip-notify

!
voice register pool 3
busy-trigger-per-button 2
id mac 0027.902A.2164
type 8851
number 1 dn 3
dtmf-relay sip-notify

!
voice register pool 4
busy-trigger-per-button 2
id mac E089.9DFB.37AD
type 8851
number 1 dn 4
dtmf-relay sip-notify
1
!
voice register pool 5
busy-trigger-per-button 2
id mac 00A2.89FA.7C88
type 8851
number 1 dn 5
dtmf-relay sip-notify
1
!
voice register pool 6
busy-trigger-per-button 2
id mac CC7F.758A.4B60
type 8851
number 1 dn 6
dtmf-relay sip-notify

!
voice register pool 7
busy-trigger-per-button 2
id mac 0027.902A.4526
type 8851
number 1 dn 7
dtmf-relay sip-notify
1
!
voice register pool 8
busy-trigger-per-button 2
id mac 0027.902A.1F94
type 8851
number 1 dn 8
dtmf-relay sip-notify

!
voice register pool 9
busy-trigger-per-button 2
id mac ACF5.E67B.D030
type 8851
number 1 dn 9
dtmf-relay sip-notify
!
voice register pool 10
busy-trigger-per-button 2
id mac 0027.902A.4D76
type 8851
number 1 dn 10
dtmf-relay sip-notify
!
!
voice translation-rule 1
rule 1 /^4/ /994/ type national national
rule 2 /^4/ /9994/ type international international
!
voice translation-rule 4
!
voice translation-rule 100
!
!
voice translation-profile SIP_OUT
translate calling 100
translate called 100
translate redirect-target 100
translate redirect-called 100
translate callback-number 100
!
!
!
license feature hseck9
license udi pid ISR4331/K9 sn FDO19171FA4
license accept end user agreement
license boot suite FoundationSuiteK9
license boot suite AdvUCSuiteK9
license boot level uck9
memory free low-watermark processor 67460
!
diagnostic bootup level minimal
!
spanning-tree mode rapid-pvst
spanning-tree extend system-id
!
username admin privilege 15 password 7 1531021F07250B757A60617745
!
redundancy
mode none
!
!
!
!
!
!
vlan internal allocation policy ascending
!
lldp run
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0/0
description *** SIP Trunk ***
ip address 10.180.12.114 255.255.255.252
load-interval 30
negotiation auto
!
interface GigabitEthernet0/0/1
no ip address
negotiation auto
!
interface GigabitEthernet0/0/2
no ip address
shutdown
media-type sfp
negotiation auto
!
interface GigabitEthernet0/1/0
description *** Connected with RTR-2 Port Gi 0/1/0 as Trunk ***
switchport trunk allowed vlan 53,110
switchport mode trunk
!
interface GigabitEthernet0/1/1
description *** Connected with FW-1 Port 8 ***
switchport trunk allowed vlan 53
switchport mode trunk
!
interface GigabitEthernet0/1/2
description *** Connected with FW-2 Port 7 ***
switchport trunk allowed vlan 53
switchport mode trunk
!
interface GigabitEthernet0/1/3
description *** Connected with WAN-SW-1 Port Gi1/0/20 ***
switchport trunk allowed vlan 52,54
switchport mode trunk
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
negotiation auto
!
interface Vlan1
no ip address
!
interface Vlan52
ip address 172.16.52.4 255.255.255.0
!
interface Vlan53
ip address 172.16.53.2 255.255.255.0
standby 53 ip 172.16.53.1
standby 53 priority 110
standby 53 preempt
load-interval 30
!
interface Vlan110
description *** SIP Backup ***
no ip address
load-interval 30
!

!
!
!
!
!
tftp-server bootflash:boot1288xx.BE-01-007.sbn
tftp-server bootflash:fbi88xx.BE-01-012.sbn
tftp-server bootflash:kern88xx.14-2-1-0001-14.sbn
tftp-server bootflash:kern288xx.14-2-1-0001-14.sbn
tftp-server bootflash:kern388xx.14-2-1-0001-14.sbn
tftp-server bootflash:m0patch288xx.BE-01-001.sbn
tftp-server bootflash:preloader88xx.BE-01-007.sbn
tftp-server bootflash:rootfs88xx.14-2-1-0001-14.sbn
tftp-server bootflash:rootfs288xx.14-2-1-0001-14.sbn
tftp-server bootflash:rootfs388xx.14-2-1-0001-14.sbn
tftp-server bootflash:sb288xx.BE-01-028.sbn
tftp-server bootflash:sb2288xx.BE-01-015.sbn
tftp-server bootflash:sb2388xx.BE-01-032.sbn
tftp-server bootflash:sip88xx.14-2-1-0001-14.loads
tftp-server bootflash:ssb288xx.BE-01-007.sbn
tftp-server bootflash:vc488xx.14-2-1-0001-14.sbn
tftp-server bootflash:/bootflash/boot1288xx.BE-01-007.sbn
!
!
control-plane
!
!
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
!
!
!
!
telephony-service
max-ephones 20
max-dn 20
ip source-address 172.16.53.2 port 2000
service dnis dir-lookup
system message CME1
time-zone 31
date-format dd-mm-yy
voicemail 100
max-conferences 10 gain -6
call-forward pattern .T
transfer-system full-blind
transfer-pattern 9.T
!
!
dial-peer voice 2 voip
description Dialpeer1
destination-pattern 1.0
session protocol sipv2
session target ipv4:172.16.53.2
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 200 voip
description *** OUTBOUND CALL TO TELECOMS ***
translation-profile outgoing SIP
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 500 voip
voice-class sip profiles 10 inbound
!
dial-peer voice 7771 voip
!
dial-peer voice 4001 voip
destination-pattern 9T
!
!
presence
presence call-list
max-subscription 200
!
sip-ua
presence enable
!
!
ephone-type 8851
device-id 8851
device-name Cisco IP Phone 8851
device-type SIP
num-buttons 1
max-presentation 1
!
!
line con 0
privilege level 15
login local
stopbits 1
line aux 0
line vty 0 4
privilege level 15
login local
length 0
transport input all
line vty 5 14
login
transport input ssh

ntp source Vlan53

!
!
!
!
!
en

what i face that internal extension working but when i configure inside to outside call that wouldn't be happen i dont know i do mistake.Moreover i dont have PVDM4 in this router so do i face an issue during configuration for outgoing calls or how do i acheive without PVDM4.

Executive Voice Config

voice service voip
mode border-element
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
trace
sip
rel1xx disable
header-passing
error-passthru
registrar server expires max 3600 min 3600
early-offer forced

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw

voice-card 0
dspfarm
dsp services dspfarm

[Roger Kallberg]

You should not need to have any PVDM module for a SIP trunk type of PSTN connection. Can you please provide the output of debug ccsip message and debug voip ccapi inout running simultaneously. Please provide the output in an attached file, not as text directly in the response as that makes it much harder to read and work with.

[abdulwadood]

Please find the logs

[Roger Kallberg]

From what I can tell there is no call in the captured logs. Did you make a call when you had enabled the debugs?

[M02@rt37]

Hello @abdulwadood,

The PVDM on a Cisco router is required when you're using hardware-based voice features, such as voice calls using Time Division Multiplexing 'TDM' or for transcoding between different voice codecs.

In your case, since you're using a SIP trunk for voice communication you may not require PVDM modules on your ISR4331 for basic voice communication.

[Roger Kallberg]

From what information are you deducting that the service provider sends those codecs? From what I could tell the debug output only contains SIP option ping and a device registration, not any actual calls with the service provider.

[M02@rt37]

I update ! Thanks.

I did not see question at end of his configuration. I'm sorry @Roger Kallberg !!!

[abdulwadood]

Thank you, Roger. I'm pleased to inform you that my issue has been resolved. Firstly, I realized that there's no need for a PVDM if you are solely using a SIP connection. Additionally, I identified the problem with the inside-to-outside and vice versa calls. It turns out that I had missed adding the proper number, including the '+' sign, in the dial peer configuration.

dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing outbound
destination-pattern .T
session protocol sipv2
session target ipv4:x.x.x.x:5060
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 21
dtmf-relay rtp-nte
no vad

dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming inbound
session protocol sipv2
session target sip-server
incoming called-number +123(DID Number)   ----->example number country code + DID number
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad

Now issue is resolved now Thank you for your assistance

[Roger Kallberg]

Glad to hear that. One advice, using destination-pattern .T on a dial peer is not recommended as it matches any dialed digits and that could result in routing loops. My advice is that you change this to be more specific, for example destination-pattern 0T if you use 0 as the brake out code, if not update to fit your specific need. Also using incoming called number to match the inbound dial peer is not recommended, your better off if you use information in the VIA header to match the inbound dial peer.

[Roger Kallberg]

Look at this document for detailed information on how call routing operates in IOS. Explain Cisco IOS and IOS XE Call Routing 

[Ibrahim AlQumairi]

Regarding the DSP and PVDM requirements: The need for DSP and PVDM modules depends on the specific requirements of your environment. If you are only using a single voice codec and do not have any transcoding or translation requirements, then you do not need to add these modules to your router. However, if you do have transcoding or translation requirements, or if you have a contact center system, then you will need to add DSP and PVDM modules to your router.

[Roger Kallberg]

Translation does not require any PVDM, transrating however does.