[asterisk-bugs] [JIRA] (ASTERISK-24602) Unable to call WebRTC client via wss on chan_pjsip
Joshua Colp (JIRA)
noreply at issues.asterisk.org
Wed Dec 20 06:29:08 CST 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-24602?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Joshua Colp updated ASTERISK-24602:
-----------------------------------
Status: Waiting for Feedback (was: Open)
Is this still a problem? WSS support has been changed and fixed, and is actively used now.
> 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, failing.log, inbound_debug (with TLS transport on).txt, pjsip.transport.conf, registration_outbound_debug(works fine).txt, working.log
>
>
> 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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