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

[Oleg Kozlov (JIRA)]

Oleg Kozlov created ASTERISK-24602:
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             Summary: 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


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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