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

Floimair Florian f.floimair at commend.com
Mon Jan 17 09:45:04 CST 2022

I don’t really see anything useful even if debug logs up to level 5.
I do have a hypothesis though.

When the call comes into Asterisk from client A it has (audio=sendrecv and video=inactive in sdp), however Asterisk starts call leg to client B with both audio and video with sendrecv.
So I guess Asterisk when it gets a reinvite with video activated from A realizes I already have video active on leg B and then does nothing on leg B.

Is there any way to “enforce” Asterisk to setup the call leg to B with the same media directions it received from leg A? I don’t mind audio automatically being sendrecv, but video is something else.


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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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