Re: DTMF to remote CUE

Heinz Schwarzfeuer · ‎07-05-2013

Hi,

I have two sites with one CME installation each, and an integrated CUE on the second site (site B).

If i´m doing a call from site A (7940 sccp phone) to the CUE via the CME of site B, i can´t get the DTMF tones through.

If i configure a dial peer on CME A which points directly to the address of the CUE in site B, the DTMF tones
are working fine.

Not-working configuration (Dial Peer points to CME on site B):

dial-peer voice 1401 voip

description To CME in Site B

destination-pattern [5][8][0-4][0-9]

session protocol sipv2

session target ipv4:10.38.0.2

codec g711ulaw

no vad

Working configuration (Dial Peer points directly to CUE on site B):

dial-peer voice 1401 voip

description To CUE in Site B

destination-pattern 5848

session protocol sipv2

session target ipv4:10.38.0.3

codec g711ulaw

no vad

CME-Configuration of Site B:

dial-peer voice 2 voip

description **Incoming Dial Peer for calls from Site A**

incoming called-number 58..

codec g711ulaw

dial-peer voice 9998 voip

description CUE PromptMgmt

destination-pattern 5848

session protocol sipv2

session target ipv4:10.38.0.3

dtmf-relay sip-notify

codec g711ulaw

no vad

!

Any hints are much appreciated!

Thanks

Heinz

Chris Deren · ‎07-05-2013

Add dtmf-relay to dial-peers on site A.

HTH,

Chris

Anas Abueideh · ‎07-05-2013

Hi Heinz,

I think it is related to dtmf negotiation, you need to configure transcoder on site B to do the dtmf negotiation.

with your non working config check debug ccsip media to see what is the negotiated dtmf.

HTH

Anas

please rate all the helpful posts

Frank Hobrecht · ‎07-05-2013

Seems to be g711 end to end, so normally there is no transcoding involved here.

asarmiento85 · ‎07-05-2013

use the following command on your dial-peer

dtmf-relay rtp-nte

this will allow the dtmf signal to travel on top of rtp

Heinz Schwarzfeuer · ‎07-06-2013

Hi,

i already tried all variants of dtmf-relays on the outgoing dial-peers of site A, but none of them worked.

It seems to be that the DTMF codes are removed from the stream on the CME router B when it forwards
the call to the CUE.

Any more ideas?

Thanks

Heinz

ashok_boin · ‎07-06-2013

Hi Heinz,

As suggested by other friend, pls share the logs captured with "debug ccsip media" command on CME at site B while making test calls to CUE.

And, have you got any logs on CmE at site A confirming the issue with DTMF only?

Ashok.

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With best regards...
Ashok

Heinz Schwarzfeuer · ‎07-07-2013

Hi,

below is the output of a "debug ccsip media" while connecting to the CUE to from Site A to Site B.

No matter what kind of DTMF relay I use on the Dial Peer of Site A, i can´t get the tones trough.

Again: If i´m pointing the dial-peer of Site A directly to the remote CUE, it always works.

Thanks

Heinz

002944: Jul 7 13:47:34.277 CEST: //-1/D79FCF7586DA/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled

002945: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002946: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 fo

r codec g711ulaw

002947: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:

Preferred Codec : g711ulaw, bytes :160

Preferred DTMF relay : inband-voice

Preferred NTE payload : 101

Early Media : No

Delayed Media : No

Bridge Done : No

New Media : No

DSP DNLD Reqd : No

002948: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002949: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIDisplayStreamInfo:

Stream type : voice-only

Media line : 1

State : STREAM_ADDING (2)

Stream address type : 1

Callid : 157399

Peer Callid : -1

RTP/SRTP Negotiated : 8

Negotiated Codec : g711ulaw, bytes :160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated DTMF relay : inband-voice

Negotiated NTE payload : 0 (tx), 0 (rx)

Negotiated CN payload : 0

Media Srce Addr/Port : [10.38.0.2]:0

Media Dest Addr/Port : [10.38.8.2]:16940

002950: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:

Stream type : voice-only

Media line : 1

State : STREAM_ADDING (2)

Stream address type : 1

Callid : 157399

Negotiated Codec : g711ulaw, bytes :160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated DTMF relay : inband-voice

Negotiated NTE payload : 0 (tx), 0 (rx)

Negotiated CN payload : 0

Media Srce Addr/Port : [10.38.0.2]:0

Media Dest Addr/Port : [10.38.8.2]:16940

002951: Jul 7 13:47:34.277 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19160 for stream 1

002952: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIDisplayStreamInfo:

Stream type : voice-only

Media line : 1

State : STREAM_ADDING (2)

Stream address type : 1

Callid : 157399

Peer Callid : -1

RTP/SRTP Negotiated : 8

Negotiated Codec : g711ulaw, bytes :160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated DTMF relay : inband-voice

Negotiated NTE payload : 0 (tx), 0 (rx)

Negotiated CN payload : 0

Media Srce Addr/Port : [10.38.0.2]:19160

Media Dest Addr/Port : [10.38.8.2]:16940

002953: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled

002954: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002955: Jul 7 13:47:34.281 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19126 for stream 1

002956: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE

002957: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: No active streams.

002958: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING

002959: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 157400) to the VOIP RTP library

002960: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002961: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1

002962: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info

laddr = 10.38.0.2, lport = 19126, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE

src_callid = 157400, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY

media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0

002963: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one

002964: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPICreateRtpSession: stun is disabled

002965: Jul 7 13:47:34.285 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp p

assthru enabled

002966: Jul 7 13:47:34.285 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=157400

002967: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002968: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around

002969: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:

Preferred Codec : g711ulaw, bytes :160

Preferred DTMF relay : sip-notify

Preferred NTE payload : 101

Early Media : No

Delayed Media : No

Bridge Done : No

New Media : No

DSP DNLD Reqd : No

002970: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002971: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIDisplayStreamInfo:

Stream type : voice-only

Media line : 1

State : STREAM_ADDING (2)

Stream address type : 1

Callid : 157400

Peer Callid : -1

RTP/SRTP Negotiated : 8

Negotiated Codec : g711ulaw, bytes :160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated DTMF relay : sip-notify

Negotiated NTE payload : 0 (tx), 0 (rx)

Negotiated CN payload : 0

Media Srce Addr/Port : [10.38.0.2]:19126

Media Dest Addr/Port : [10.38.0.3]:20898

002972: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: 0 Active Streams

002973: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!

002974: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo:

caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=

002975: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=

002976: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)

002977: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:

Stream type : voice-only

Media line : 1

State : STREAM_ADDING (2)

Stream address type : 1

Callid : 157400

Negotiated Codec : g711ulaw, bytes :160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated DTMF relay : sip-notify

Negotiated NTE payload : 0 (tx), 0 (rx)

Negotiated CN payload : 0

Media Srce Addr/Port : [10.38.0.2]:19126

Media Dest Addr/Port : [10.38.0.3]:20898

002978: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:35CCDBD8

002979: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING

002980: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 157399) to the VOIP RTP library

002981: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002982: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1

002983: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info

laddr = 10.38.0.2, lport = 19160, raddr = 10.38.8.2, rport=16940, do_rtcp=TRUE

src_callid = 157399, dest_callid = 157400, stream type = voice-only, stream direction = SENDRECV

media_ip_addr = 10.38.8.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0

002984: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one

002985: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPICreateRtpSession: stun is disabled

002986: Jul 7 13:47:34.325 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp p

assthru enabled

002987: Jul 7 13:47:34.325 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=157399

002988: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIGetNewLocalMediaDirection:

New Remote Media Direction = SENDRECV

Present Local Media Direction = SENDRECV

New Local Media Direction = SENDRECV

retVal = 0

002989: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING

002990: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 157400) to the VOIP RTP library

002991: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2

002992: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1

002993: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info

laddr = 10.38.0.2, lport = 19126, raddr = 10.38.0.3, rport=20898, do_rtcp=TRUE

src_callid = 157400, dest_callid = 157399, stream type = voice-only, stream direction = SENDRECV

media_ip_addr = 10.38.0.3, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0

002994: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update

002995: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:35CCDBD8

002996: Jul 7 13:47:34.325 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=157400

002997: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIGetNewLocalMediaDirection:

New Remote Media Direction = SENDRECV

Present Local Media Direction = SENDRECV

New Local Media Direction = SENDRECV

retVal = 0

002998: Jul 7 13:47:45.077 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:367D9708

002999: Jul 7 13:47:45.077 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:35CCDBD8

003000: Jul 7 13:47:45.081 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIHandleDestroyRtpSession: stream:367D9708

003001: Jul 7 13:47:45.081 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIHandleDestroyRtpSession: stream:35CCDBD8

avinsrid · ‎07-07-2013

Hi Heinz,

On the inbound dial peer of CME B can you please configure session protocol sipv2 and try the dtmf?

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Regards, Avinash

asarmiento85 · ‎07-07-2013

Also I would say add the dtmf relay on the inbound dialpeer.
Remember that this dial peer is forwarding to your cue module.
I hope it helps.

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