WebRTC to SIP Calling

[virtualtalk]

1. Brekeke Product Name and Version: PBX 3.12.2.2

2. Java version: 11

3. OS type and the version: Win 2012

4. UA (phone), gateway or other hardware/software involved: Cisco AS5350

5. Your problem:

Hi,

I am trying to make a call from a webpage using jssip through Brekeke PBX (actually tried sip server first but docs seemed to say I need pbx) to cisco AS5400 and out to PSTN.

So far:

I have set up pbx with auth, certificate and turned on wss/ws/tls. I also made phone type webrtc.

1. Web client can register under registered clients
2. I use the following dial plan to try to make the call:

$request = ^INVITE
To = sip:(phone number I want to call)@

To = sip:%1@ip address of cisco box:5060
$transport = UDP

3. I can see the call attempts but the Cisco returns an error saying SDP doesn't match up. Brekeke reports Error 400..

4. I feel as though I should be converting WebRCT to SIP somewhere so the router understands? Is that right, how do I do that?


Many thanks!

[redroof]

If you want to make call from a WebRTC client such as JsSIP, a callee client must support ICE and DTLS.
So you can make call from JsSIP to another JsSIP without any issues.
But most of non-WebRTC SIP clients such as Cisco can not accept a call sent from WebRTC.

Please refer to the pages below. Brekeke PBX can convert SDP and connect WebRTC client to/from non-WebRTC client.
https://docs.brekeke.com/interop/webrtc-client
https://docs.brekeke.com/pbx/setting-up ... ser-webrtc