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

[Thomas Arimont (JIRA)]

Thomas Arimont created ASTERISK-28617:
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             Summary: 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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