[asterisk-bugs] [JIRA] (ASTERISK-29839) res_pjsip: Failover occurs after CANCEL

[Jonas Swiatek (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29839?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=257671#comment-257671 ] 

Jonas Swiatek edited comment on ASTERISK-29839 at 1/7/22 7:19 AM:
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A feature request:

Adding a configuration option, that will cause PJSIP to never fail over if it's received any sort of 100-199 range provisional response.

In our setup, we have a set of servers further downstream (kamailio), which handles failover. What I really want, is for Asterisk to only failover if it completely fails to establish any sort of dialog with.


was (Author: jonasswiatek):
A feature request:

Adding a configuration option, that will cause PJSIP to never fail over it's received any sort of 100-199 range provisional response.

In our setup, we have a set of servers further downstream (kamailio), which handles failover. What I really want, is for Asterisk to only failover if it completely fails to establish any sort of dialog with.

> res_pjsip: Failover occurs 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: Unassigned
>         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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