Cisco CUBE + SBC + DTMF issues

Unanswered Question
Aug 13th, 2010
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Hello all, I'm having an issue interconnection a CUBE system with a Session Border Controller (Audiocodes/nCite Netrake).


According to the Cisco side:



Regarding the toll free number issue I see that the issue with the carrier sending the SDP


with G711ulaw only instead of G729r8 and G711ulaw on the domestic and long distance calls.



From my scope (Session Border Controller Engineer), I don't change any codecs, it's a straight passthrough. It would be foolish of me to set a static "729" followed by X since all clients differ (some use Cisco, some use Avaya, some use Asterisk) therefore my trunks are ALWAYS set up on a broad configuration. I've never experienced this particular problem with codec issues since *that* control is NOT on my end. What a client does after we pass the call through is on them.



Below is a sanitized config and since I'm unfamiliar with CUBE, CUCM, I'd figure I'd ask the experts. Is there anything in this config that can be changed?



voice call send-alert
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
address-hiding
allow-connections sip to sip
no supplementary-service sip moved-temporarily
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
h323
sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  header-passing error-passthru
  midcall-signaling passthru
  g729 annexb-all
  sip-profiles 100
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice class h323 1
  h225 timeout tcp establish 3
!
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
!
!
voice translation-rule 1000
rule 1 /^1503.../ //
!
voice translation-rule 2000
rule 1 /^9/ //
!
!
voice translation-profile CUCM-OUTBOUND
translate called 1000
translate redirect-called 1000
!
voice translation-profile CUSTOMER--PROVIDER-OUTBOUND
translate called 2000
translate redirect-called 2000
!
!
interface GigabitEthernet0/0
description Voice VLAN
ip address 192.168.1.1 255.255.255.0
!
dial-peer voice 1000 voip
description ** Inbound Peer **
voice-class codec 1
fax-relay ecm disable
no vad
!
dial-peer voice 1010 voip
description ** Outbound 4-digit to CUCM **
translation-profile outgoing CUCM-OUTBOUND
destination-pattern 1503.......
voice-class codec 1
session protocol sipv2
session target ipv4:<IP Address of CUCM Subscriber>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!
dial-peer voice 1011 voip
description ** Outbound 4-digit to CUCM **
translation-profile outgoing CUCM-OUTBOUND
preference 1
destination-pattern 1503.......
voice-class codec 1
session protocol sipv2
session target ipv4:<IP Address of CUCM Publisherr>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!
dial-peer voice 9000 voip
description ** Outbound Emergency Services to CUSTOMER **
translation-profile outgoing CUSTOMER--PROVIDER-OUTBOUND
destination-pattern 9911
voice-class codec 1
voice-class sip profiles 100
session protocol sipv2
session target ipv4:<IP Address of SIP Provider>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!
dial-peer voice 9001 voip
description ** Outbound Emergency Services to CUSTOMER **
destination-pattern 911
voice-class codec 1
voice-class sip profiles 100
session protocol sipv2
session target ipv4:<IP Address of SIP Provider>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!
dial-peer voice 9010 voip
description ** Outbound Non-emergency Services to CUSTOMER **
translation-profile outgoing CUSTOMER--PROVIDER-OUTBOUND
destination-pattern 9[2-8]11
voice-class codec 1
voice-class sip profiles 100
session protocol sipv2
session target ipv4:<IP Address of SIP Provider>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!
dial-peer voice 9020 voip
description ** Outbound 11-digit to CUSTOMER **
translation-profile outgoing CUSTOMER--PROVIDER-OUTBOUND
destination-pattern 91[2-9].........
voice-class codec 1
voice-class sip profiles 100
session protocol sipv2
session target ipv4:<IP Address of SIP Provider>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!
dial-peer voice 9030 voip
description ** Outbound International to CUSTOMER **
translation-profile outgoing CUSTOMER--PROVIDER-OUTBOUND
destination-pattern 9011T
voice-class codec 1
voice-class sip profiles 100
session protocol sipv2
session target ipv4:<IP Address of SIP Provider>
session transport udp
dtmf-relay rtp-nte
fax-relay ecm disable
no vad
!

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carunach Fri, 08/13/2010 - 08:20
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  • Cisco Employee,

Configuration looks fine.


What is the exact problem? DTMF not recognized / not detected / clipped / dual dtmf?


Couple of things to check :


1) Does RTP-NTE get negotiated for the SIP call? Are the right set of inbound and outbound dial-peers matched? ("debug voip ccapi inout" and "debug ccsip message" will provide this information)


2) Where is the call terminated?


3) Does the remote endpoint / phone support RTP-NTE?


4) We can turn on "debug voip rtp session named-event" in the CUBE to see the type of DTMF received and sent.


Arun

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