Summary: | ASTERISK-24271: Unable to make WebRTC call through chan_PJSIP nor chan_SIP | ||
Reporter: | Dafi Ni (dafi) | Labels: | |
Date Opened: | 2014-08-26 09:08:31 | Date Closed: | 2014-09-15 07:01:52 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | 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/Linux | Attachments: | ( 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. |