| Summary: | ASTERISK-24725: WebSockets uses first loaded SIP module(chan_sip, pjsip) as SIP provider for WebRTC | ||||
| Reporter: | Aleksei Kulakov (Each) | Labels: | |||
| Date Opened: | 2015-01-27 02:26:14.000-0600 | Date Closed: | 2015-01-27 05:38:58.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Resources/res_http_websocket Resources/res_pjsip_transport_websocket | ||
| Versions: | 12.4.0 13.1.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Ubuntu 14.04, Asterisk 13.1.0, pjproject(github.com/asterisk/pjproject 21/jan/2015) | Attachments: | |||
| Description: | 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] | ||||
| Comments: | |||||