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Tiberie Kirijas
Beginner

Send DTMF digits with in-band metod

Hi,

Can anyone suggest what Cisco DTMF method can be configured under the Dial-Peer that will transmit the digits within the g.7xx audio i.e. pure in-band DTMF payload ?

Regards,

Tiberie

7 REPLIES 7

If you want to do inband within rtp stream, using codec payloads, then you dont need to configure anything at all, however this is not reliable..If no DTMF relay is configured then the gateway willd efault to inband (ie passing dtmf digits within the rtp stream)

NB: This is different from RTP-NTE where dtmf tones are transmitted within rtp stream but in a different payload (97-126)

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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Nadeem Ahmed
Cisco Employee

What signalling protocol are you using ? I mean SIP, H323 or MGCP ?

According to that you would need to configure dtmf..


Br,
Nadeem 

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Br, Nadeem Please rate all useful post.

Nadeem,

I'm using SIP signaling protocol on a CUBE router towards SP provider.

CUCM<----SIP---->CUBE<----SIP---->SP.

Tiberie

If you are using sip, why do you want to do inband then? You should be doing rtp-nte and that is configured on each dial-peer as ff:

dial-peer voice 1 voip

dtmf-relay rtp-nte

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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The service provider SIP platform supports the DTMF standard under RFC2833  or in other words, pure in-band (DTMF tones in the actual audio stream). So, in some situation when the call goes to certain AA/IVR systems, the relayed DTMF tones are not relayed and properly recognized by the other side.

I've read that "rtp-nte" is not actualy doing in-band transmition of the DTFM tones, rather the digits are transmitted via data, not in-the-audio of the g.7xx codec.

That's why, I am asking if there is a way to configure the SIP dial-peers on the CUBE to send the compatible DTMF tones to the SP.

Thank You both for your assistance.

p.s.

At the moment, "dtmf-relay rtp-nte" is configured but this does not work for some AA/IVR's.

Regards,

Tiberie

rtp-nte is RFC2833 and transmits dtmf tones in the audio stream..It doesnt use the codec payload, it uses rtp payload range 97-126..The most common payload used is 101....

You will need to troubleshoot why DTMF is failing to those IVRs..It is possible they dont understand the dtmf tones been sent to them..To do this...you shoulle enable the ff debugs

debug ccsip messages

debug voip rtp session named-event

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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An UPDATE:

I used the commands:"dtmf-interworking standard" under the two dial-peers (cucm/cube & cube/SP) and the DTMF tones were transmitted/received fine from then onwards. The SIP SP is accepting only In-band tones for telephony events i.e. as part of the voice waveform. See http://tools.ietf.org/html/rfc4733#

While the command "dtmf rtp-nte" is sending telephony event out-of-band but within the RTP session, which is not in reality an in-band method.

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