Summary: | ASTERISK-22645: Broad media offers from Jitsi client results in a crash in ast_copy_pj_str at res_pjsip.c | ||
Reporter: | Rusty Newton (rnewton) | Labels: | |
Date Opened: | 2013-10-03 09:24:14 | Date Closed: | 2013-10-03 09:54:01 |
Priority: | Critical | Regression? | |
Status: | Closed/Complete | Components: | Resources/res_pjsip Resources/res_pjsip_nat |
Versions: | SVN 12.0.0-alpha1 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | SVN-branch-12-r400356 | Attachments: | ( 0) backtrace_jitsi_call1.txt ( 1) full_jitsi_call1.txt |
Description: | Reproduction:
1. Register Jitsi SIP account to Asterisk with default settings, except username, host IP and password. 2. Make a call from the Jitsi SIP account to an Asterisk extension. Looks like it crashes on receiving the INVITE I suspect it is combination of misconfiguration in NAT related settings and one of the media offers from Jitsi: {noformat} ÿv=0^M ÿo=6002 0 0 IN IP4 127.0.0.1^M ÿs=-^M ÿc=IN IP4 127.0.0.1^M ÿt=0 0^M ÿm=audio 5005 RTP/AVP 96 9 97 98 100 102 0 8 103 3 104 101^M ÿa=rtpmap:96 opus/48000^M ÿa=fmtp:96 usedtx=1^M ÿa=rtpmap:9 G722/8000^M ÿa=rtpmap:97 SILK/24000^M ÿa=rtpmap:98 SILK/16000^M ÿa=rtpmap:100 speex/32000^M ÿa=rtpmap:102 speex/16000^M ÿa=rtpmap:0 PCMU/8000^M ÿa=rtpmap:8 PCMA/8000^M ÿa=rtpmap:103 iLBC/8000^M ÿa=rtpmap:3 GSM/8000^M ÿa=rtpmap:104 speex/8000^M ÿa=rtpmap:101 telephone-event/8000^M ÿa=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level^M ÿm=video 5007 RTP/AVP 105 99^M ÿa=recvonly^M ÿa=rtpmap:105 H264/90000^M ÿa=fmtp:105 profile-level-id=4DE01f;packetization-mode=1^M ÿa=imageattr:105 send [x=[0-640],y=[0-480]] recv [x=[0-1920],y=[0-1080]]^M ÿa=rtpmap:99 H264/90000^M ÿa=fmtp:99 profile-level-id=4DE01f^M ÿa=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1920],y=[0-1080]]^M {noformat} As changing settings in Jitsi, to result in the below offer, then works fine with no crash: {noformat} v=0 o=6002 0 0 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 5013 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level {noformat} Or, alternatively, leaving the default offers in Jitsi and adding a "localnet=127.0.0.1" line to my transport config also resulted in no crash. {noformat} [transport-udp-nat] type=transport protocol=udp bind=0.0.0.0 localnet=192.168.1.0/24 localnet=127.0.0.1 external_media_address=1.2.3.4 external_signaling_address=1.2.3.4 {noformat} | ||
Comments: |