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2/23/2021 3:09 PM
 

I would like to know what I can't do if I choose to use only webRTC as my engine. As I was testing, when I receive a call in the webphone and try to hold the call, it would send a different call-id from what I had received from the invite, so it couldn't signal the call to hold.

Another thing is that when I end the call on a SIP, the webphone gets the BYE signal, however, it doesn't end the call and don't trigger the onCallStateChange event.

I couldn't get conference to work too, with webRTC the only way for conference to work is to have the server handle it? Because none of the conference parameters worked for me.

Using Kamailio and asterisk

Thanks!

 
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