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

[cervajs (JIRA)]

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

cervajs updated ASTERISK-24602:
-------------------------------

    Comment: was deleted

(was: with defined tls transport in pjsip.conf it show like asterisk wants dial webrtc_cervenka with tls transport. but webrtc_cervenka is defined with transport wss

-- Executing [6001 at from_log:1] Dial("SIP/vr1a885-00000001", "PJSIP/webrtc_cervenka") in new stack
[Sep  1 14:16:17] DEBUG[7156]: rtp_engine.c:425 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xb6d73294'
[Sep  1 14:16:17] DEBUG[7156]: res_rtp_asterisk.c:2489 ast_rtp_new: Allocated port 18080 for RTP instance '0xb6d73294'
    -- Called PJSIP/webrtc_cervenka
[Sep  1 14:16:17] DEBUG[7156]: rtp_engine.c:434 ast_rtp_instance_new: RTP instance '0xb6d73294' is setup and ready to go
[Sep  1 14:16:17] DEBUG[7156]: res_rtp_asterisk.c:4721 ast_rtp_prop_set: Setup RTCP on RTP instance '0xb6d73294'
[Sep  1 14:16:17] DEBUG[7156]: sdp_srtp.c:122 ast_sdp_crypto_alloc: local_key64 or/bMXpXveEnxJtspA5fcfNdGvUa2V2wGQvuSktE len 40
[Sep  1 14:16:17] DEBUG[7156]: res_pjsip_sips_contact.c:57 sips_contact_on_tx_request: Upgrading contact URI on outgoing SIP request to SIPS due to SIPS Request URI
[Sep  1 14:16:19] DEBUG[7166]: threadpool.c:1107 worker_idle: Worker thread idle timeout reached. Dying.
[Sep  1 14:16:23] DEBUG[7095]: chan_sip.c:8765 __sip_alloc: Allocating new SIP dialog for 6ddb91c07a42e2814e2bbcea0ba13db1 at 213.168.163.68:5060 - OPTIONS (No RTP)
[Sep  1 14:16:27] DEBUG[7160]: threadpool.c:1107 worker_idle: Worker thread idle timeout reached. Dying.
[Sep  1 14:16:30] ERROR[7085]: pjsip:0 <?>:      tlsc0x9fcd904 TLS connect() error: Connection timed out [code=120110]
[Sep  1 14:16:30] WARNING[7085]: pjsip:0 <?>:    tsx0xb6d023d4 Failed to send Request msg INVITE/cseq=28639 (tdta0x9fc0378)! err=120110 (Connectio        
)

> 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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