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

[Joeran Vinzens (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28634?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249710#comment-249710 ] 

Joeran Vinzens commented on ASTERISK-28634:
-------------------------------------------

while trying to reproduce the issue the logs are telling exactly the same in the test and in production. You cannot see anything regarding the re-invites except the pjsip logger

Both log say the call gets established, both enter the bridge and change the bridge type to native and at the end of the call bridge changes back and call gets hung up.

> 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
>            Assignee: Unassigned
>         Attachments: 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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