[asterisk-bugs] [JIRA] (ASTERISK-22645) Broad media offers from Jitsi client results in a crash in ast_copy_pj_str at res_pjsip.c

Rusty Newton (JIRA) noreply at issues.asterisk.org
Thu Oct 3 09:25:03 CDT 2013


     [ https://issues.asterisk.org/jira/browse/ASTERISK-22645?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-22645:
------------------------------------

    Attachment: full_jitsi_call1.txt
                backtrace_jitsi_call1.txt
    
> Broad media offers from Jitsi client results in a crash in ast_copy_pj_str at res_pjsip.c
> -----------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-22645
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22645
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip, Resources/res_pjsip_nat
>    Affects Versions: SVN, 12.0.0-alpha1
>         Environment: SVN-branch-12-r400356
>            Reporter: Rusty Newton
>            Severity: Critical
>         Attachments: backtrace_jitsi_call1.txt, full_jitsi_call1.txt
>
>
> 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}

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