[asterisk-bugs] [JIRA] (ASTERISK-29051) res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29051?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-29051:
-------------------------------------

    Target Release Version/s: 13.38.0

> res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used
> ---------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-29051
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29051
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp
>    Affects Versions: 17.6.0
>         Environment: Debian Buster
>            Reporter: Sebastian Damm
>            Assignee: Unassigned
>            Severity: Minor
>      Target Release: 13.38.0
>
>         Attachments: alltraffic.pcap, asterisk.log, asterisk-nocolor.log
>
>
> When bridging two call legs where one leg supports rfc4733 events and the other leg does not, DTMF tones don't get converted. This is because Asterisk enters native bridge if codecs are equal and then has no chance to detect anything inside the rtp stream. When transcoding from one codec to another, Asterisk stays in simple bridge, and it should behave the same way if dtmf modes differ. 
> To reproduce: Set dtmf_mode to "auto" in the endpoint settings in pjsip.conf. Send a call from a client only supporting inband DTMF to the Asterisk, send this call to another client supporting telephone-events. Then send DTMF digits from the calling device. They will end up inband on the receiving client. However, if the receiving client is for example another Asterisk, it will not look into the audio if the SDP offered telephone-event. DTMF digits will not be recognized.



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