[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:comment-tabpanel&focusedCommentId=227436#comment-227436 ] 

cervajs commented on ASTERISK-24602:
------------------------------------

after patching pjsip to ignore sips we moved to the next problem
call from webrtc_cervenka to webrtc_tomec
webrtc_tomec answered call but channel hangup
any ideas howto debug why hangup happened?

[Sep 2 11:25:28] DEBUG[28870] res_pjsip_session.c: Source of transaction state change is RX_MSG 
[Sep 2 11:25:28] DEBUG[28870] res_pjsip_session.c: Received response [Sep 2 11:25:28] DEBUG[28870] res_pjsip_session.c: Response is 200 OK [Sep 2 11:25:28] DEBUG[28870] pjsip: inv0xb6c2acd4 ....Got SDP answer in Response msg 200/INVITE/cseq=5501 (rdata0xb6c679dc) 
[Sep 2 11:25:28] DEBUG[28870] pjsip: inv0xb6c2acd4 ....SDP negotiation done, status=220046 
[Sep 2 11:25:28] DEBUG[28870] pjsip: inv0xb6c2acd4 ....Received Response msg 200/INVITE/cseq=5501 (rdata0xb6c679dc), sending ACK 
[Sep 2 11:25:28] DEBUG[28870] pjsip: endpoint ....Request msg ACK/cseq=5501 (tdta0xb6c0b188) created. 
[Sep 2 11:25:28] DEBUG[28870] pjsip: dlg0xb6c2acd4 .....Sending Request msg ACK/cseq=5501 (tdta0xb6c0b188) 
[Sep 2 11:25:28] DEBUG[28594] threadpool.c: Increasing threadpool stasis-core's size by 1 
[Sep 2 11:25:28] DEBUG[28899][C-0000000f] channel.c: Hanging up channel 'PJSIP/webrtc_tomec-00000008'

> 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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