[asterisk-bugs] [JIRA] (ASTERISK-29839) PJSIP Failover after cancel

[Jonas Swiatek (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29839?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jonas Swiatek updated ASTERISK-29839:
-------------------------------------

    Attachment: bla.pcap

Relevant parties in this pcap

Asterisk Server: 172.28.32.24
Kamailio Server 1: 172.28.21.119
Kamailio Server 2: 172.28.20.196

The relevant parties are the invites going FROM asterisk TO the two kamailio servers.

The first invite, from Asterisk -> Kamailio 2 is great, but as it's seen in the trace, kamailio doesn't return it's 487 Request Terminated response immediate, and 10 seconds in, AFTER asterisk has cancelled, it sends out the same invite to Kamailio 1 which isn't supposed to happen.

> PJSIP Failover after cancel
> ---------------------------
>
>                 Key: ASTERISK-29839
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29839
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 16.24.0
>         Environment: Amazon Linux 2
>            Reporter: Jonas Swiatek
>            Assignee: Jonas Swiatek
>         Attachments: bla.pcap, debug_log_dnssrv
>
>
> Asterisk configured with an AOR like so:
> [registrar]
> type=aor
> contact=somesrvrecord.example.com
> somesrvrecord.example.com returns multiple IP addresses.
> The issue I'm seeing is that Asterisk will perform a failover, even after it has cancelled an outbound call. For instance:
> 1. Asterisk sends a SIP INVITE
> 2. Asterisk sends a SIP CANCEL to cancel that invite
> 3. If asterisk receives anything but 200 cancelled followed by 487 Request Terminated, it will STILL failover by sending the same INVITE to the next IP address in the SRC Record set.
> This obviously creates a ghost call, and if that new INVITE is answered, asterisk immediately generates a BYE to end that transaction, which I suppose is fine, but it shouldn't be failing over to the next record under these conditions at all.
> This also happens if it doesn't receive the 487 Request Terminated within 10 seconds.
> Generally asterisk is pretty aggressive with it's failovers here. Almost anything but a 404, 486 or 480 will cause a failover, which could be a bug, but at least would be very nice if could be configured to only do a failover on either a 500 response or a timeout on the initial SIP INVITE.



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