[asterisk-bugs] [JIRA] (ASTERISK-28634) Invite loop within PJSIP

Joshua C. Colp (JIRA) noreply at issues.asterisk.org
Mon May 18 09:11:25 CDT 2020


     [ https://issues.asterisk.org/jira/browse/ASTERISK-28634?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp updated ASTERISK-28634:
--------------------------------------

    Attachment: ASTERISK-28871.diff

Please try the attached patch. This returns the behavior to that of past versions. Improved codec negotiation is occurring for Asterisk 18, and this functionality could be revisited then to be improved.

> Invite loop within PJSIP
> ------------------------
>
>                 Key: ASTERISK-28634
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28634
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 17.0.0
>         Environment: Asterisk 17.0.0 with bundled PJSIP on Debian Buster
>            Reporter: Joeran Vinzens
>              Labels: patch
>      Target Release: 17.1.0
>
>         Attachments: ASTERISK-28871.diff, asterisk.conf, asterisk_invite_loop_other_side.pcap, invite_loop_small.pcap, pjsip.conf, rtp.conf, Screenshot from 2019-11-26 10-19-59.png, sipp_scenario.tar.xz
>
>
> Fom time to time we can see traffic increase on out Asterisk Systems. After some debugging we see the Asterisk send kind of an Invite loop over and over. Since there is no way to deny re-invites in PJSIP there is no way out of it. From asterisk perspective it sends these re-invites as fast as it can which result in an increase of 3MBit/s of signalling traffic from this maschine. So far we do not know how to reproduce the issue but we captured the scenario from our production system. (trace attached) the Asterisk log itself just tells us the Invite is going out but nothing further. There was no trigger on the other call-leg.



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