[asterisk-bugs] [JIRA] (ASTERISK-24271) Unable to make WebRTC call through chan_PJSIP nor chan_SIP

Joshua Colp (JIRA) noreply at issues.asterisk.org
Wed Aug 27 08:25:28 CDT 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24271?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222423#comment-222423 ] 

Joshua Colp commented on ASTERISK-24271:
----------------------------------------

The crashes mentioned here have been fixed in 12 SVN and will be in the next release.

> Unable to make WebRTC call through chan_PJSIP nor chan_SIP
> ----------------------------------------------------------
>
>                 Key: ASTERISK-24271
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24271
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Channels/chan_sip/General
>    Affects Versions: 12.5.0
>         Environment: Linux KRA-WS-DAFI 3.13.0-24-generic #46-Ubuntu SMP Thu Apr 10 19:11:08 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux
>            Reporter: Dafi Ni
>         Attachments: 12.5.0-chan_pjsip-udp2ws-debug.log, 12.5.0-chan_pjsip-udp2wss-debug.log, 12.5.0-chan_pjsip-ws2udp-debug.log, 12.5.0-chan_pjsip-ws2ws-backtrace.log, 12.5.0-chan_pjsip-ws2ws-debug.log, 12.5.0-chan_pjsip-wss2udp-debug.log, 12.5.0-chan_pjsip-wss2wss-debug.log, 12.5.0-chan_sip-udp2ws-dabug.log, 12.5.0-chan_sip-udp2wss-debug.log, 12.5.0-chan_sip-ws2udp-debug.log, 12.5.0-chan_sip-ws2ws-debug.log, 12.5.0-chan_sip-wss2udp-debug.log, 12.5.0-chan_sip-wss2wss-debug.log, pjsip.conf, sip.conf
>
>
> Calls with DTLS-SRTP to DTLS-SRTP and sip (UDP) always failed, no matter that are on WS, WSS, UDP or using chan_sip or chan_pjsip
> I have configured it with ssl keys and this configuration is working on 11.11.0 (except ASTERISK-24146 )
> on chan_pjsip with WS to WS calls i got segfault (backtrace attached)
> I have made logs from different options.
> _PJSIP summary:_
> *ws, wss -> udp* (ringing, after anwser)
> {quote}
> ERROR: pjsip:0 <?>: 	icess0x7fe4880 ..Error sending STUN request: Invalid argument
> WARNING: res_rtp_asterisk.c:1667 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on '0x7fe48c025d50'
> WARNING: res_rtp_asterisk.c:3944 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.
> {quote}
> *udp -> wss* (not ringing)
> {quote}
> WARNING: pjsip:0 <?>: 	tsx0x7fe48c053 ...Failed to send Request msg INVITE/cseq=2117 (tdta0x7fe48c00a170)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
> {quote}
> *udp -> ws* (ringing, segfault )
> {quote}
> asterisk: ../src/pjsip/sip_resolve.c:351: pjsip_resolve: Assertion `!"Unknown transport type"' failed.
> {quote}
> *wss ->wss* (not ringing)
> {quote}
> WARNING: pjsip:0 <?>: 	tsx0x7f9a44013 ...Failed to send Request msg INVITE/cseq=20228 (tdta0x7f9a1c003f70)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
> {quote}
> *ws -> ws* (ringing, segfault )
> {quote}
> asterisk: ../src/pjsip/sip_resolve.c:351: pjsip_resolve: Assertion `!"Unknown transport type"' failed.
> {quote}



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