Hi there,
I am using WebSocket transport to create a WebRTC application. I can see the SDP data in the PJSIP logger, but I cannot retrieve it via the Node ari-client
.
Is it possible to get the SDP data?
Hi there,
I am using WebSocket transport to create a WebRTC application. I can see the SDP data in the PJSIP logger, but I cannot retrieve it via the Node ari-client
.
Is it possible to get the SDP data?
No, it is not. The most you can do is get a little bit of information using the CHANNEL dialplan function[1], specifically IP address/port.
How do I work on a WebRTC audio call application? How do I connect a WebRTC client application?
Can I use only SIP information for this, or do I need additional configurations?
The supported signaling protocol for WebRTC with Asterisk is SIP. If you don’t want to use SIP, then there is nothing else directly built-in so you would need to implement it yourself in C, or use an external intermediary that converted from whatever you want to use to SIP.
okay thank you
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