One reinvite problem


#1

hello,every body.
I has one problem, so i ask for help here.

I use asterisk to transfer webrtc to my sip gateway. but my sip gateway don’t support webrtc. so I use asterisk.

I find this work well, when i call one video or two video .

but when a call has set up. sip gateway add a new video (change one video to two video). the other one don’t receive invite.

Can i change asterisk configure to support this?

I find some community topic say asterisk is b2b model don’t tranfer reivite.

some body can tell me answer。 thinks。


#2

Asterisk 16 may support this but it hasn’t really been tested for that use case. What version are you using?


#3

I use asterisk 16.0.1.


#4

If it doesn’t work out of the box, then it would likely require someone to specifically go through the scenario, identify what has to happen, and code it.


#5

thank you.
because I don’t has ability to code asterisk. i change my plan.

may be i can add some funny function to asterisk.