Higher sample rates for telephone-event

Greetings!

I was wondering if there is any way to use higher sample rate frequencies for DTMF, as this seem to be a requirement?

We got reports from some partners that DTMF failed when connecting with OPUS. After some investigating of the SIP-messages we saw that the incoming INVITE contained telephone-event lines both for 8000 and 48000, but in the response from Asterisk (18.13.0) only a line with 8000 is present in the SDP. This seem to break the DTMF handling for the calling party.

After googleing a bit, I found some posts refering to problems of this kind, like:

https://groups.google.com/g/discuss-webrtc/c/LntxAkr2H_U

which refers to:

saying that here it is stated that one must use the same frequency.

I also found an old post from the asterisk-dev mailing list that asks about the same issue:

https://asterisk-dev.digium.narkive.com/eEcODzly/telephone-event-at-rates-other-than-8000

The response from Joshua Colp indicates that there is no support for this, but it is a quite old posting so things may have changed.

So what is the current status? Is there a way to handle this currently? If not, should this be regarded as a bug? (If the above postings are correct when claiming it is mandatory).

Best regards,
Torbjörn Abrahamsson

It’s not supported currently. There is a pull request up[1] to add support.

[1] rtp_engine: add support for multirate RFC2833 digits by mbradeen · Pull Request #700 · asterisk/asterisk · GitHub

Thank you for the quick reply!

I will take a look at the code, and see if it is possible to use on Asterisk 18 (as I guess the pull request is for the latest version).

Once again, thank you!

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