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Inband Transcoding Question:

philjaeger
Beginner
Beginner

We are using inband dtmf for some of our SIP endpoints, which won't work with Cisco Unity.   We tried to inject some hardware transcoders but it doesn't appear to work.   My first question is,  can you transcode an inband dtmf tone to out of band?  Is the transcoder smart enough to detect the tone in an rtp stream?    If it can then I must have a configuration issue.   

My next question is, will Unity Connection support inband dtmf?

10 Replies 10

yuanwu
Cisco Employee
Cisco Employee

Hi, Philijaeger

The Cisco UBE will support RFC2833 to in-band voice DTMF using transcoding resource, and it only support the G711 codec at the call leg with in-band voice dtmf , and you need to register the transcoding resource to the local CUBE as the call agent.

Unity Connection will support RFC2833 and OOB DTMF, not in-band voice DTMF.

Rgds/Randy

Thank you for the information but I have a follow up question.  You stated CUBE supports rfc2833 to inband (G711) but does it support  inband (G711) to rfc2833?   We have a need to convert inband dtmf to oob.    Thanks again for the response. 

I think Cisco Unity (Out of band DTMF) to Cisco SIP end point (RFC2833-Inband) should work with the help of transcoder.

Hope the following tech note may help...

http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1015823

Regards...

-Ashok.


With best regards...
Ashok

Thanks for the post but the issue is not calling from Unity to the SIP endpoint but vice versa.  When our SIP endpoint calls into Unity we need to convert inband to out of band.    Are you suggesting that this document will also help me with the call scenario I provided?   Thanks again.

Hello!

We are having the same problem; did you encounter any solution?

Our carrier is sending us DTMF with voice inband, so we need to convert this to RFC2833 for Unity compatibility.

According to http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html -see table 3- this should be feasible by using a transcoder.

Otherwise, can you point me to a Cisco document that explaings DTMF relay conversion capabilities?

Thanks!

We ended up using transcoding with our non-Cisco session border controllers to correct the issue.  We could not get it to work with our Cisco gateways.

 

Sorry I couldn't be more help.

 

Regards.

philjaeger
Beginner
Beginner

Anyone else have transcoding advice?

Hi,

Cisco Unity supports RFC2833 in-band DTMF natively which is supported by SIP endpoints as well. Do you have any specific reason to use in-band DTMF for SIP endpoints instead of RFC 2833?

I have found another informative link below...

https://lists.cs.columbia.edu/pipermail/sip-implementors/2007-June/016970.html

Regards...

-Ashok.


With best regards...
Ashok

The SIP endpoint has another issue with using RFC2833.  (Talk-off)  so we had to use inband dtmf.   We were hoping to use a transcoder to convert the inband tones to out of band but so far have been unsuccessful.  I don't believe the IOS hardware transcoders can detect inband tones and convert them...at least we haven't been able to do it.   SIP info is also not supported by our carrier. 

Thanks.

Phil

david.alfaro1
Beginner
Beginner

Well, I have a similar question. If a provider ITSP is sending raw in-band DTMF into g711 to inbound dial-peer; it is well known that it is problematic when using compressed audio codec in the outbound dial-peer. It seems when CuBe does sip2sip interworking, cube is able to receive raw in-band DTMF with g711, and then CUBE can convert this raw in-band to rtp-nte as long as CUBE LTI transcoder is available. In order of that I would guess configuring inbound dial-peer with codec g711 and NO dtmf-relay configuration (becausd it assumes the default DTMF relay configuration as raw in-band dtmf audio). And then I guess I am supposed to configure a outbound dial-peer with any other codec for example g729 and dtmf-relay rtp-nte, targeting a CUC (assuming CUC configured to support g729). 

In order of the above, is that correct? If no, how do you customize your inbound or outbound dial-peers in such scenario ?

Unfortunatelly there are ITSPs yet offering raw in-band dtmf audio =/

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