DTMF not working and RTP is not streaming

[khaleelmd51]

Hi Team,

I am new here and this is my first post in the community,

I am having an issue with Cisco Gateway with DTMF and RTP. I have configured the dial-peers as per the service provider requirement. 

but the DTMF and RTP are not working. any assistance on how to start troubleshooting about checking if the RTP and DTMF are reaching Cisco Gateway from the Service provider.

Regards

Mohammed

[VON CLAWSON]

That would depend on your configuration. You haven't mentioned if this is a SIP carier and you are doing SIP trunks or if it is a PRI or what. If it is SIP then you need to make sure that you have dtmf-relay rtp nte on the dial-peer. What you are looking for from the carrier is in the SIP invite in the SDP section. You are looking for a line similar to this one

m=audio 19838 RTP/AVP 0 101

The 0 indicates g.711ulaw and the 101 indicates DTMF RFC2833 or rtp nte

[Nithin Eluvathingal]

Can you share configuration you made and the ISP requirement.

[khaleelmd51]

Dears Thanks a lot for the response, kindly find the below configuration details.

IN dial-peer voice 203 VoIP i have configured a manual RTP type to 116 because SP was saying that the payload was sending different. so still it is the same issue but for dial-peer voice 101 VoIP is the same as mentioned.

ART_CC_SIP_GW#sh run
Building configuration...

Current configuration : 3432 bytes
!
! Last configuration change at 02:24:38 UTC Wed May 5 2021 by root
!
version 16.9
service timestamps debug datetime msec
service timestamps log datetime msec
platform qfp utilization monitor load 80
no platform punt-keepalive disable-kernel-core
!
hostname ART_CC_SIP_GW
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
enable secret 5 $1$bake$pVlvvzAko6aDq2B/zM1sV0
!
no aaa new-model
!
!
!
!
login on-success log
!
!
!
!
!
!
!
subscriber templating
multilink bundle-name authenticated
!
!
!
!
!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 10.80.92.13
ipv4 10.80.92.16
mode border-element license capacity 10
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
sip
!
!
!
!
!
voice register pool-type SIP
!
!
!
!
!
voice-card 0/4
no watchdog
!
license udi pid ISR4331/K9 sn FDO19230EJ7
license accept end user agreement
no license smart enable
diagnostic bootup level minimal
!
spanning-tree extend system-id
!
!
!
username root password 0 XXXXX
!
redundancy
mode none
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0/0
ip address 10.189.131.150 255.255.255.252
negotiation auto
!
interface GigabitEthernet0/0/1
ip address 10.80.92.111 255.255.255.0
negotiation auto
!
interface GigabitEthernet0/0/2
no ip address
negotiation auto
!
interface Service-Engine0/4/0
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
negotiation auto
!
ip forward-protocol nd
no ip http server
no ip http secure-server
ip tftp source-interface GigabitEthernet0
ip default-network 10.189.131.0
ip route 0.0.0.0 0.0.0.0 10.170.17.193
ip route 0.0.0.0 0.0.0.0 GigabitEthernet0/0/0
ip route 10.141.40.233 255.255.255.255 10.189.131.149
!
!
!
!
!
!
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-conferences 8 gain -6
transfer-system full-consult
!
!
dial-peer voice 203 voip
description ####Transfering Calls From VGW to the Main CIC IPPBX####
destination-pattern 8386350
rtp payload-type cisco-codec-ilbc 99
rtp payload-type nte 116
session protocol sipv2
session target ipv4:10.80.92.13:5060
session transport udp
voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3
codec g711alaw
no vad
!
dial-peer voice 201 voip
description ####Transfering Calls From VGW to the Main CIC IPPBX####
preference 1
destination-pattern 8386300
session protocol sipv2
session target ipv4:10.80.92.13
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 101 voip
description ####Outgoing calls Through STC SIP Trunk####
destination-pattern .
session protocol sipv2
session target ipv4:10.141.40.233:5060
session transport udp
voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3
dtmf-relay rtp-nte
codec g711alaw
no vad
!
!
!
line con 0
transport input none
stopbits 1
line aux 0
stopbits 1
line vty 0 4
password XXXXXX
login local
transport input ssh
line vty 5
password XXXXXX
login local
transport input ssh
line vty 6 15
login local
transport input ssh

Let me if you i need to provide more details.

Regards

Mohammed

[VON CLAWSON]

This is good information. 

I'm wondering about these in the 203 dial-peer

rtp payload-type cisco-codec-ilbc 99
rtp payload-type nte 116

Can you send the output from debug ccsip messages so we can see the invite and how the carrier is responding?

Thanks

[khaleelmd51]

HI Von,

I have added the log file, kindly have a look and your assistance will be appreciated highly,

Regards

[Nithin Eluvathingal]

Hope you are in the Saudi, can you try the below mentioned DTMF.

https://community.cisco.com/t5/telepresence-and-video/saudi-telecom-sip-configuration/td-p/2829354

[khaleelmd51]

Hi, Nithin,

I have tried this as well, still no success

Regards

[Nithin Eluvathingal]

Try the below. shutdown dial-peer 203.

dial-peer voice 203 voip
Shutdown
!
dial-peer voice 201 voip
description ####Transfering Calls From VGW to the Main CIC IPPBX####
preference 1
destination-pattern 8386300
session protocol sipv2
session target ipv4:10.80.92.13
session transport udp
dtmf-relay rtp-nte 
codec g711alaw
no vad
!
dial-peer voice 101 voip
description ####Outgoing calls Through STC SIP Trunk####
destination-pattern .
session protocol sipv2

rtp payload-type nte 116
session target ipv4:10.141.40.233:5060
session transport udp
voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3
dtmf-relay rtp-nte 
codec g711alaw
no vad
!

[khaleelmd51]

Hi Nithin,

Thanks for the update,

Actually, I have tried this command and getting the below error

ART_CC_SIP_GW(config-dial-peer)#rtp payload-type nte 116
ERROR: value 116 in use!

Regards

[Nithin Eluvathingal]

On which dial-peer you tried this ?

[khaleelmd51]

I tried on both ones,

Firstly I have deleted the 203 dial-peer and shutdown the 201 dial-peer 

first I tried this command on 101 then the 201 dial-peer

I can see that this command is not working in any dial-peer. I think this payload is using by another service as below. I used 

GW#sh dial-peer voice 201

RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, iSAC=124
MP4A-LATM=111, lmr_tone=0, nte_tone=0
h263+=118, h264=119
G726r16 using static payload
G726r24 using static payload
RTP comfort noise payload type = 19

Regards

[khaleelmd51]

Hi,

One strange thing I found, I am not sure if it is correct or not but for me, it looks strange. the thing is if I am removing the Outbound dial-peer the incoming calls are not working. it gives as network busy if I am dailing the DID number. 

Just wanted to understand is it normal behavior?

Regards