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

[Alexander Traud (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29051?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=252034#comment-252034 ] 

Alexander Traud commented on ASTERISK-29051:
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Anyone working on this already? I face a similar symptom with native bridge and consider to look into it. My issue has a complete different cause as it is not about different DTMF methods but a compatible but [different audio codec …|https://github.com/traud/asterisk-amr/issues/10] Therefore, an idea for a short-term fix: Is it possible to instruct Asterisk to transcode always, go for simple bridge instead? Not aware of. Anyone interested in that approach, as a short-term fix?

> 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
>         Attachments: alltraffic.pcap, asterisk.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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