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

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29051?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=251828#comment-251828 ] 

Joshua C. Colp edited comment on ASTERISK-29051 at 8/28/20 6:10 AM:
--------------------------------------------------------------------

The problem is not with the bridging layer, but res_pjsip_sdp_rtp. It does not appear to be setting the correct information on the RTP instance when "auto" DTMF is used resulting in the logic that determines DTMF compatibility believing they are compatible.


was (Author: jcolp):
The problem is not with the bridging layer, but res_pjsip_sdp_rtp. It does not appear to be setting the correct information on the RTP instance when "auto" DTMF is used.

> 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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)