DTMF issue different payload and clock rate

[Guillermo_PY]

Hello Team, I need your help!

I have a problem with the DTMF tone in this scenario:

ITPS>>>CUBE>>>SIPtrunk>>>CUCM>>>SipTrunk>>>UnityConnection

I got this message from my provider in my CUBE:

Jun 24 09:17:23.050: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+11112222@net.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.14.0.141:5060;branch=z9hG4bK1x94rbzo4bzzb4r0bryxb7390T00371
Call-ID: isbcxw7ab3004wy0779zi7ooi0byixr7wy01@B.5.102.net.com
From: "+555566666"<sip:+555566666@net.com;noa=national;srvattri=national>;tag=sbc0511w5sg0rxp
To: <sip:+11112222@10.14.3.194:5060;transport=udp;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,BYE,CANCEL,UPDATE,INFO,PRACK,NOTIFY,REFER,SUBSCRIBE,OPTIONS,MESSAGE
P-Charging-Vector: icid-value=DFE7801000-0624-09174700;orig-ioi=net.com;term-ioi=net.com
Max-Forwards: 65
Supported: timer,100rel,histinfo,in-band-dtmf,early-session
Session-Expires: 1800
Min-SE: 600
P-Asserted-Identity: <tel:+555566666>
Contact: <sip:10.14.0.141:5060;Dpt=eb6a-200>
Content-Length: 232
Content-Type: application/sdp
Content-Disposition: session
v=0
o=- 707023 707023 IN IP4 10.14.0.142
s=SBC call
c=IN IP4 10.14.0.142
t=0 0
m=audio 31948 RTP/AVP 8 18 0 108
a=fmtp:18 annexb=yes
a=rtpmap:108 telephone-event/16000
a=fmtp:108 0-15
a=ptime:20
a=maxptime:40
a=sendrecv

I don't understand why I receive this 16000 clock rate

Guillermo

[Guillermo_PY]

I add more information to try to understand my problem:

GW-CM#debug ccsip info
SIP Call info tracing is enabled
GW-CM#
GW-CM#debug ccsip feature dtmf
dtmf debugging for ccsip info is enabled (active)
GW-CM#
GW-CM#
GW-CM#
GW-CM#ter
GW-CM#terminal moni
GW-CM#terminal monitor
GW-CM#
GW-CM#
GW-CM#
*Jun 24 18:25:42.322: //54/6AE20A0B803D/SIP/Info/info/2080/sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list!
*Jun 24 18:25:42.322: //54/6AE20A0B803D/SIP/Info/info/32/sipSPIStreamTypeAndDtmfRelay: DTMF Relay mode: Inband Voice
*Jun 24 18:25:42.322: //-1/xxxxxxxxxxxx/SIP/Info/sipSPI_Check_If_ICE_Needed: CANDIDATE attribute, level 1not found.
*Jun 24 18:25:42.322: //54/6AE20A0B803D/SIP/Info/notify/33/sipSPIHandleInviteMedia:
Negotiated Codec : g711alaw, bytes :160
Preferred Codec : g711alaw, bytes :160
Preferred DTMF relay 1 : 0
Preferred DTMF relay 2 : 0
Negotiated DTMF relay : 0
Preferred and Negotiated NTE payloads: 98 0
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated V150.1 Modem Passthrough: 0 0
Preferred and Negotiated V150.1 Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0

*Jun 24 18:25:42.323: //54/6AE20A0B803D/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 0,
*Jun 24 18:25:42.325: //54/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.325: //55/6AE20A0B803D/SIP/Info/info/32/sipSPI_ipip_set_pld_flags_from_ex_caps: Peer cap provided: dtmf = 0, t38 version = 0 t38 maxBitRate = 14400
*Jun 24 18:25:42.325: //55/6AE20A0B803D/SIP/Info/info/32/sipSPIDtmfTranscoder: local codec 6, peer codec 6
*Jun 24 18:25:42.325: //55/6AE20A0B803D/SIP/Info/info/32/sipSPIDtmfTranscoder: local DTMF 6, peer DTMF 0
*Jun 24 18:25:42.325: //55/6AE20A0B803D/SIP/Info/info/288/sipSPI_sip_CheckAndReserveTranscoder: need transcoding for dtmf mismatch
*Jun 24 18:25:42.325: //55/6AE20A0B803D/SIP/Info/notify/304/sipSPICodecTranscoder: Looks like DTMF/ SRTP/ Volume adj needs xcoder So set OFFER_ALL to true, & send all the codecs in Initial INVITE..
*Jun 24 18:25:42.326: //55/6AE20A0B803D/SIP/Info/critical/288/sipSPI_sip_CheckAndReserveTranscoder: Xcoder reservation failed for dtmf. Dont disconnect the call.
*Jun 24 18:25:42.326: //55/6AE20A0B803D/SIP/Info/verbose/32/sipSPIAddSDPPayloadAttributes: DTMF max_event 16
*Jun 24 18:25:42.326: //55/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.327: //55/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.328: //55/6AE20A0B803D/SIP/Info/critical/32/sipSPIUpdateCallEntry:
DTMF interworking cases are not support digit display
*Jun 24 18:25:42.343: //55/6AE20A0B803D/SIP/Info/ccsipIsMediaForkingByeConsume: Entry to ccsipIsMediaForkingByeConsume
*Jun 24 18:25:42.343: //55/6AE20A0B803D/SIP/Info/ccsipIsMediaForkingByeConsume: Exit from ccsipIsMediaForkingByeConsume
*Jun 24 18:25:42.343: //55/6AE20A0B803D/SIP/Info/sipSPIDeferCallClose: Entering here
*Jun 24 18:25:42.344: //54/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.345: //56/6AE20A0B803D/SIP/Info/info/32/sipSPI_ipip_set_pld_flags_from_ex_caps: Peer cap provided: dtmf = 0, t38 version = 0 t38 maxBitRate = 14400
*Jun 24 18:25:42.345: //56/6AE20A0B803D/SIP/Info/info/32/sipSPIDtmfTranscoder: local codec 6, peer codec 6
*Jun 24 18:25:42.345: //56/6AE20A0B803D/SIP/Info/info/32/sipSPIDtmfTranscoder: local DTMF 6, peer DTMF 0
*Jun 24 18:25:42.345: //56/6AE20A0B803D/SIP/Info/info/288/sipSPI_sip_CheckAndReserveTranscoder: need transcoding for dtmf mismatch
*Jun 24 18:25:42.345: //56/6AE20A0B803D/SIP/Info/notify/304/sipSPICodecTranscoder: Looks like DTMF/ SRTP/ Volume adj needs xcoder So set OFFER_ALL to true, & send all the codecs in Initial INVITE..
*Jun 24 18:25:42.345: //56/6AE20A0B803D/SIP/Info/critical/288/sipSPI_sip_CheckAndReserveTranscoder: Xcoder reservation failed for dtmf. Dont disconnect the call.
*Jun 24 18:25:42.346: //56/6AE20A0B803D/SIP/Info/verbose/32/sipSPIAddSDPPayloadAttributes: DTMF max_event 16
*Jun 24 18:25:42.346: //56/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.347: //56/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.347: //56/6AE20A0B803D/SIP/Info/critical/32/sipSPIUpdateCallEntry:
DTMF interworking cases are not support digit display
*Jun 24 18:25:42.443: //56/6AE20A0B803D/SIP/Info/info/32/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
*Jun 24 18:25:42.443: //-1/xxxxxxxxxxxx/SIP/Info/sipSPI_Check_If_ICE_Needed: CANDIDATE attribute, level 1not found.
*Jun 24 18:25:42.444: //56/6AE20A0B803D/SIP/Info/info/32/sipSPI_ipip_upd_2833_dtmf_params: setting ipip_caps DTMF to RFC2833: callid = 56, dtmf = 6
*Jun 24 18:25:42.444: //56/6AE20A0B803D/SIP/Info/critical/32/sipSPIUpdateCallEntry:
DTMF interworking cases are not support digit display
*Jun 24 18:25:42.444: //54/6AE20A0B803D/SIP/Info/info/32/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB
*Jun 24 18:25:42.445: //54/6AE20A0B803D/SIP/Info/info/32/sipSPI_ipip_set_pld_flags_from_ex_caps: Peer cap provided: dtmf = 6, t38 version = 0 t38 maxBitRate = 14400
*Jun 24 18:25:42.445: //54/6AE20A0B803D/SIP/Info/info/32/sip_iwf_sip_copy_audio_channelInfo_to_stream: DTMF 101 SilenceSuppression 1
*Jun 24 18:25:42.445: //54/6AE20A0B803D/SIP/Info/critical/32/sipSPIUpdateCallEntry:
DTMF interworking cases are not support digit display
*Jun 24 18:25:42.446: //56/6AE20A0B803D/SIP/Info/notify/32/ccsip_query_dtmf_nte_pt_info: Negotiated dtmf rx: 101, tx: 101
*Jun 24 18:25:42.446: //54/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob disabled
*Jun 24 18:25:42.446: //54/6AE20A0B803D/SIP/Info/notify/32/ccsip_bridge:
set dtmf_iw_enabled to FALSE
*Jun 24 18:25:42.446: //56/6AE20A0B803D/SIP/Info/notify/2080/sipSPIUpdateRtcpSession: DTMF inb/oob iwf enabled 0
*Jun 24 18:25:42.446: //56/6AE20A0B803D/SIP/Info/notify/32/ccsip_bridge:
DTMF inb/oob iwf enabled 0
*Jun 24 18:25:53.478: //54/6AE20A0B803D/SIP/Info/ccsipIsMediaForkingByeConsume: Entry to ccsipIsMediaForkingByeConsume
*Jun 24 18:25:53.478: //54/6AE20A0B803D/SIP/Info/ccsipIsMediaForkingByeConsume: Exit from ccsipIsMediaForkingByeConsume
*Jun 24 18:25:53.480: //54/6AE20A0B803D/SIP/Info/sipSPIDeferCallClose: Entering here
*Jun 24 18:25:53.491: //56/6AE20A0B803D/SIP/Info/sact_disconnecting_new_message_response: [sact_disconnecting_new_message_response] Received Response Class [2] Method Code [103]

*Jun 24 18:25:53.492: //56/6AE20A0B803D/SIP/Info/sipSPIDeferCallClose: Entering here

[b.winter]

Hi,

"I don't understand why I receive this 16000 clock rate" --> There is nothing to understand. ITSP is sending the DTMF method like that, and that's all. You can complain about that, but the provider won't change anything (because he probably needs to change his complete SIP platform).

You just can react to it.

[Guillermo_PY]

@b.winter thank you, I understand

[Roger Kallberg]

AFAICT you have not actually explained your actual problem that you might have. Can you please share a little more information on this? One thing to check is that you have No Preference set as the DTMF setting on both your trunks, both the one from CM to SBC for your service provider connection and from CM to CUC for the integration with Unity Connection.

[Guillermo_PY]

@Roger Kallberg hello.

Follow my diagram with the configuration that I have:

ITSP>>>siptrunk>>>CUBE>>>siptrunk>>>CUCM>>>UnityConnection>>>CallHandler

Cube Incoming from ITSP dial-peer with rtp-nte, Outgoing to CUCM dial-peer with rtp-nte, on CUCM Sip trunk to CUBE with MTP enable and dtmf No preference, on CUCM Sip trunk to Unity with MTP enable and dtmf No preference and SIP trunk on Unity with RFC2833 and KPML enable.

[Roger Kallberg]

I’m sorry but this does not actually outline whatever problem that you have. Can you please clearly describe what problem that you’re facing?

[Roger Kallberg]

One note. On the dial peer to/from CM to your SBC the recommendation from Cisco is to have both rtp-nte and kpml DTMF enabled on the dial peers in both directions.

[Guillermo_PY]

@Roger Kallberg I configured LTI and the DTMF work now.

https://www.youtube.com/watch?v=TwXTpgdjOeA