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.

FLORIAN FLOIMAIR
Development

Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
commend.com
LG Salzburg / FN 178618z


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


CAUTION: This email originated from outside of the organization. Do not click 
links or open attachments unless you recognize the sender and know the content 
is safe.
On Wed, Jan 12, 2022 at 10:57 AM Floimair Florian 
<f.floim...@commend.com<mailto:f.floim...@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
Check us out at 
www.sangoma.com<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.sangoma.com%2F&data=04%7C01%7Cf.floimair%40commend.com%7C4b1ff012a9ae4026dc0d08d9d5dc4a55%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637775964444664699%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000&sdata=rrgZ0cvdPh%2FWJ%2ByuY1bJGVOp8e8P%2FAKGxf5J3ucHlGQ%3D&reserved=0>
 and 
www.asterisk.org<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.asterisk.org%2F&data=04%7C01%7Cf.floimair%40commend.com%7C4b1ff012a9ae4026dc0d08d9d5dc4a55%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637775964444664699%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000&sdata=GG1W28r%2FF%2F97YK7Kdp2yeAcfAzHWUQV1WezSrq7VhE0%3D&reserved=0>
-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to