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

Aleksei Kulakov (JIRA) noreply at issues.asterisk.org
Tue Jan 27 02:27:34 CST 2015


Aleksei Kulakov created ASTERISK-24725:
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             Summary: 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: 13.1.0, 12.4.0
         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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