Can we use same FreePBX Extension for WebRTC and PJSIP calling

[kiranpk (Kiran)]

Hi Team , Can we use same FreePBX Extension for WebRTC and PJSIP calling is it possible.

[dwsiemens (Dan S)]

Yes, You can do use both the technology is sip over websockets.

[kiranpk (Kiran)]

Can you please tell me the procedure or what setting needs to do.
thanks in advance

[dwsiemens (Dan S)]

so at the very basic,
Create you pjsip extension. make sure you have selected the webrtc and avap stuff.

Depending on how you decide to implement you have to decide if you are doing sip over websockets. or trying to connect internally I provide traffic over the internet so I use Session Border controllers so in my configs I set the SBC to forword internet traffic to my asterisk instance. I listen externally on ports 8089 for websockets convert to sip internal, and ports 5061 for TLS and pass along. So in my case I can user SRTP…

If not that way you have to use DTLS to support Websockets and your Sip clients can’t be secured to TLS. Only tcp or udp. I don’t know of any sip clients that support DTLS. Webrtc has to be DTLS based on RFC.

I really recommend a SBC. They don’t have to be so expensive to break the bank but they will save you from a set of bad choices. I use the Jssip webrtc phone client. But I have also looked at sipJS like in Round pin. and others.