WebRTC dSIPRouter v0.78 + Vitalpbx

[MASTERING VOIP]

Hi Mack Hendricks,

Please confirm, can it be implemented in dsiprouter as shown in the image below?
I watched your tutorial video on YouTube
https://www.youtube.com/watch?v=nOHwrmuLLL0
However, the tutorial uses FusionPBX as the backend.



Below, I'll explain our needs.

Kamailio acts as a WebRTC gateway and will be passed through if I register using the WebRTC client (JSSIP).

The protocol from JSSIP to Kamailio uses WSS + TLS, while from Kamailio to VitalPBX (base on Asterisk) uses SIP.

All extensions will be present in VitalPBX, for example, PJSIP extensions 4001-4005.

From the JSSIP client, I can use these extensions to register with dsiptouer, which will then be forwarded to VitalPBX, so that the extension status in VitalPBX will be registered/reachable.


For the call test, the scenario is:

4001 registers using the JSSIP client to Kamailio,
4002 registers to VitalPBX using the MicroSIP client.

4001 can make calls to 4002, and vice versa, and from 4001, calls to the telco provider can be made.

Does Dsiprouter v0.78 support this by default, or do I need a subscription?

Could you please create a tutorial for WebRTC Dsiprouter + VitalPBX so I can see that this actually works.

Thanks

[Mack Hendricks]

Yes, this use case works.  This use case would require the core subscription, which is $200/month or $2400 a year.  Our support would help you through the implementation process.

I will put this use case in the queue for a video.  

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[MASTERING VOIP]

Thank you for your answer. Can I have a one-month subscription?

What do I need to prepare for testing? Do I need to set up two servers in the cloud? dsiprouter and vitalpbx, complete with their domains pointed to dsiprouter and vitalpbx?

Thanks