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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29147?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=252611#comment-252611 ] 

Asterisk Team commented on ASTERISK-29147:
------------------------------------------

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> 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
>              Labels: webrtc
>
> 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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