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Summary:ASTERISK-24271: Unable to make WebRTC call through chan_PJSIP nor chan_SIP
Reporter:Dafi Ni (dafi)Labels:
Date Opened:2014-08-26 09:08:31Date Closed:2014-09-15 07:01:52
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_pjsip Channels/chan_sip/General
Versions:12.5.0 Frequency of
Occurrence
Constant
Related
Issues:
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/LinuxAttachments:( 0) 12.5.0-chan_pjsip-udp2ws-debug.log
( 1) 12.5.0-chan_pjsip-udp2wss-debug.log
( 2) 12.5.0-chan_pjsip-ws2udp-debug.log
( 3) 12.5.0-chan_pjsip-ws2ws-backtrace.log
( 4) 12.5.0-chan_pjsip-ws2ws-debug.log
( 5) 12.5.0-chan_pjsip-wss2udp-debug.log
( 6) 12.5.0-chan_pjsip-wss2wss-debug.log
( 7) 12.5.0-chan_sip-udp2ws-dabug.log
( 8) 12.5.0-chan_sip-udp2wss-debug.log
( 9) 12.5.0-chan_sip-ws2udp-debug.log
(10) 12.5.0-chan_sip-ws2ws-debug.log
(11) 12.5.0-chan_sip-wss2udp-debug.log
(12) 12.5.0-chan_sip-wss2wss-debug.log
(13) backtrace-3.09.2014.txt
(14) pjsip.conf
(15) sip.conf
Description: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}

Comments:By: Joshua C. Colp (jcolp) 2014-08-27 08:24:12.549-0500

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

By: Rusty Newton (rnewton) 2014-08-29 18:17:03.781-0500

[~dafi] Can you please test with the latest 12 SVN version and let us know which issues remain for you?

By: Dafi Ni (dafi) 2014-09-03 09:01:37.037-0500

Now with svn version of branch 12, i have another segfault with chan_pjsip. (backtrace-3.09.2014.txt)

* Connect peer and register  (ws or wss) with chrome and jssip,
* Without unregister make page reload
* At next register attempt segfault exist


By: Joshua C. Colp (jcolp) 2014-09-03 09:13:25.495-0500

I believe I've fixed it in SVN. Please retry.

By: Dafi Ni (dafi) 2014-09-15 05:22:41.183-0500

Now with Svn version is OK.