[asterisk-dev] [External] Re: How to debug reinvites not getting forwarded to other call leg (pjsip)

Floimair Florian f.floimair at commend.com
Wed Jan 12 09:07:09 CST 2022

Thanks, I will dig deeper in there.

Symphony Cloud Services (1568)

Von: asterisk-dev <asterisk-dev-bounces at lists.digium.com> im Auftrag von "Joshua C. Colp" <jcolp at sangoma.com>
Antworten an: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Datum: Mittwoch, 12. Jänner 2022 um 16:00
An: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Betreff: Re: [asterisk-dev] [External] Re: How to debug reinvites not getting forwarded to other call leg (pjsip)

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On Wed, Jan 12, 2022 at 10:57 AM Floimair Florian <f.floimair at commend.com<mailto:f.floimair at commend.com>> wrote:
Hi Joshua!

No it does not concern audio in this case, it is a change in media for video. Initially the call is established with a=recvonly (or even a=inactive) that then changes to a=sendrecv (when the camera is activated on the linphone).

The debug log should tell you what is going on then with the handling of the streams. It's spread across res_pjsip_sdp_rtp, res_pjsip_session, and then the bridge modules. I don't have specific line numbers because there's lots and lots involved. The "core show channel" CLI command can also provide insight.

Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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