| Summary: | ASTERISK-24639: Crash with PJSIP on SIP to SIP over WebSockets call (WebRTC, SIPML5) | ||||
| Reporter: | Rusty Newton (rnewton) | Labels: | |||
| Date Opened: | 2014-12-22 16:23:05.000-0600 | Date Closed: | |||
| Priority: | Major | Regression? | No | ||
| Status: | Waiting for Feedback/In Progress | Components: | Resources/res_rtp_asterisk | ||
| Versions: | Frequency of Occurrence | Constant | |||
| Related Issues: |
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| Environment: | * Asterisk SVN-branch-13-r429983 * PJPROJECT 2.3 Compiled from source with (./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-speex --with-external-srtp --with-external-gsm CFLAGS='-O2 -DNDEBUG -DPJ_HAS_IPV6=1'), * OpenSSL 1.0.1-4ubuntu5.20 | Attachments: | ( 0) backtrace.txt ( 1) extensions.txt ( 2) full.txt ( 3) http.txt ( 4) jssip_full.txt ( 5) pjsip.txt ( 6) rtp.txt | ||
| Description: | Seemingly very similar to ASTERISK-24334, except happens when using PJSIP, newer openssl, newer PJPROJECT and Asterisk 13 as well.
h1. Reproduction To reproduce, I just follow the tutorial that worked in the past: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 The crash happens when calling from a SIP phone to the WebRTC client. In this case, a Digium D40 to SIPML5 (live demo). h1. Notes backtrace.txt is the trace from the crash occurring when calling from a Digium D40 to SIPML5. The full.txt is the full log trace with pjsip logger output. The jssip_full.txt contains a full log from the same call scenario, but swapping out the SIPML5 client with JsSIP. Calling from the D40 to JsSIP results in a failed call, but no crash. JsSIP responds to our INVITE with 488 Not Acceptable Here. | ||||
| Comments: | By: Rusty Newton (rnewton) 2014-12-22 17:02:54.767-0600 Attaching config files used. By: Sean Bright (seanbright) 2018-09-17 15:23:57.746-0500 Is this reproducible with current Asterisk 13? By: Sean Bright (seanbright) 2021-08-10 14:39:14.157-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-dev on the IRC network irc.libera.chat to reopen the issue should you have the additional information requested. Further information on issue tracker usage can be found in the Asterisk Issue Guidelines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||||