[asterisk-bugs] [JIRA] (ASTERISK-24602) Unable to call WebRTC client via wss on chan_pjsip

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24602?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24602:
------------------------------------

    Assignee: Oleg Kozlov
      Status: Waiting for Feedback  (was: Triage)

Oleg can you provide an Asterisk log (gathered following these instructions:https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information) that shows the complete working and failing calls. 

Be sure that SIP debug is enabled. What we want to see if is all the various logging channels such as VERBOSE and DEBUG integrated with the SIP trace.

> Unable to call WebRTC client via wss on chan_pjsip
> --------------------------------------------------
>
>                 Key: ASTERISK-24602
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24602
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.0.0
>         Environment: Centos 6.5 x86
> pjproject 2.3 (https://github.com/asterisk/pjproject)
>            Reporter: Oleg Kozlov
>            Assignee: Oleg Kozlov
>         Attachments: endpoint_config.txt, inbound_debug (with TLS transport on).txt, pjsip.transport.conf, registration_outbound_debug(works fine).txt
>
>
> Calls to WebRTC client (sipml5) via WSS transport or chan_pjsip always fail.
> Registration and calls from WebRTC client work without issues.
> I believe that the issue is about Asterisk trying to use wrong transport (TLS instead of WSS) to SIP INVITE WebRTC clients.
> Error summary:
> 1. TLS transport isn't configured in pjsip.conf:
> bq. pjsip:0 <?>: 	 tsx0xb740c914 ...Failed to send Request msg INVITE/cseq=5413 (tdta0xb740d3c8)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
> 2. TLS transport is configured for some other peers in pjsip.conf:
> {quote}
> 	<--- Transmitting SIP request (1741 bytes) to TLS:CLIENT_IP:62950 --->
> 	INVITE sips:tempwss2 at CLIENT_IP:62950;transport=wss;rtcweb-breaker=no SIP/2.0
> 	...
> 	pjsip:0 <?>: tlsc0x884bf24 TLS connect() error: Connection timed out
> {quote}



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