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| 00:19.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 03:01.05 | lonepurplesmurf | How are conversations encrypted with asterisk? Any good links on this? |
| 03:01.07 | lonepurplesmurf | thanks a lot |
| 03:01.23 | lonepurplesmurf | does it use tls and then aes for bulk encrypting? |
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| 04:13.04 | drmessano | SRTP is pretty standard |
| 04:13.08 | drmessano | So just google for it |
| 04:13.11 | drmessano | Oh he left |
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| 06:55.22 | jkroon | so there seems to be some interesting things around websocket transports, ws vs wss, both transport types in pjsip.conf requires a bind address which leads me to believe they open dedicated ports, but still it seems they rely on the built-in http server (which I'll probably end up wrapping using apache's mod_proxy_ws). https://pastebin.com/c1J6MqrP represents my config. ports 6062 and 6063 remains closed even though "pjsip show transports" lists them. |
| 06:55.22 | jkroon | <PROTECTED> |
| 06:55.49 | jkroon | is it possible to have res_http_websocket prefer pjsip for sip? |
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