[asterisk-bugs] [JIRA] (ASTERISK-24725) WebSockets uses first loaded SIP module(chan_sip, pjsip) as SIP provider for WebRTC

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24725?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp closed ASTERISK-24725.
----------------------------------

    Resolution: Duplicate

> WebSockets uses first loaded SIP module(chan_sip, pjsip) as SIP provider for WebRTC
> -----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24725
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24725
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_http_websocket, Resources/res_pjsip_transport_websocket
>    Affects Versions: 12.4.0, 13.1.0
>         Environment: Ubuntu 14.04, Asterisk 13.1.0, pjproject(github.com/asterisk/pjproject 21/jan/2015)
>            Reporter: Aleksei Kulakov
>
> When using both chan_sip(for everything except for WebRTC) and PjSIP(for WebRTC only) there is no way to customize which one would be used as SIP provider for WebRTC. 
> In my case chan_sip is alway selected by default, which is clearly not what i want. In this case module res_pjsip_transport_websocket displayed as {{'Not running'}} in {{'module show'}} output.
> Modifying modules.conf preload section is only way to workaround this behavior that i've found.
> There it is:
> {code:title=modules.conf|borderStyle=solid}
> [modules]
> autoload=yes
> preload => res_sorcery_astdb.so
> preload => res_sorcery_memory.so
> preload => res_sorcery_config.so
> preload => res_http_websocket.so
> preload => res_pjsip.so
> preload => res_pjsip_outbound_publish.so
> preload => res_pjsip_pubsub.so
> preload => res_pjsip_session.so
> preload => res_pjsip_transport_websocket.so
> ;... remaining contents of default modules.conf
> {code}
> With this hack res_pjsip_transport_websocket registers itself earlier than chan_sip and thus making pjsip stack responsible for handling WebRTC connecitons.
> *We need some other way configure this without messing with modules.conf * Some config option for both chan_sip & pjsip that disables registration in res_http_websocket or WS/WSS url customisation option would be ok
> This bug is sibling of [ASTERISK-24106]



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