[asterisk-bugs] [JIRA] (ASTERISK-28617) DTMF over SIP INFO in scenarios without audio does not work well

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28617?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248744#comment-248744 ] 

Asterisk Team commented on ASTERISK-28617:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

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> DTMF over SIP INFO in scenarios without audio does not work well
> ----------------------------------------------------------------
>
>                 Key: ASTERISK-28617
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28617
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/Channels
>    Affects Versions: 13.21.1
>         Environment: Linux, openembedded
>            Reporter: Thomas Arimont
>            Severity: Critical
>
> (actually affected version Asterisk-Certified 13.21-cert3)
> The problem is essentially the same as in ASTERISK-28245.
> Besides the direct media scenario we have an additionally scenario where a special DATUS SIP client is involved. This device mutes audio by default in transmit direction (no rtp frames) and activates audio only by a foot switch. In this situation dtmf input (pin for conferences , transfer features codes , etc) using SIP INFO mode is not understood properly especially when SIP INFO are sent quick after one another.
> I attach a patch which is working so far. But we need verification/improvement.
> The patch is also working if successive dtmf events are received quicker than the signalled event duration (plus minimum gap/pause) allows, i.e. DTMF events has to be buffered in the ast channel read queue and emulation has to be processed asynchronously at slower speed.



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