[asterisk-bugs] [JIRA] (ASTERISK-29147) Chan PJSIP + WebRTC + Chan SIP Setup.

[Hitesh K (JIRA)]

Hitesh K created ASTERISK-29147:
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             Summary: Chan PJSIP + WebRTC + Chan SIP Setup.
                 Key: ASTERISK-29147
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29147
             Project: Asterisk
          Issue Type: Information Request
      Security Level: None
          Components: Channels/chan_pjsip, Channels/chan_sip/General, Resources/res_http_websocket
    Affects Versions: 16.8.0
         Environment: OS: Ubuntu 18.04 Server Image
            Reporter: Hitesh K
            Severity: Blocker


Hi there,

We are experiencing a strange issue with the WebSocket (for WebRTC) modules. When the chan_sip is loaded, the WebSocket module stop running, and when we unload the chan_sip module, the WebSocket modules work perfectly.

On the other note, we have working endpoints for traditional sip channels with PJSIP configurations. Now, we would like to enable webRTC for those endpoints as well so that any endpoint can be connected through either a WebSocket client or a sip client (softphone). 

We have created separate transports for both TLS and WSS, and individually they work fine, but when we try adding in the "|webrtc=yes" to the endpoint, it no longer works with the SIP channels (TLS  - Softphone) and visa-versa for WebRTC.

Is there a way possible for us to be able to configure an endpoint to work with both webRTC and PJSIP channels through a browser-based webRTC client and a softphone, respectively?

Thanks



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