[asterisk-bugs] [JIRA] (ASTERISK-24239) UDP sockets used for RTP are sometimes leaked

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24239?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan closed ASTERISK-24239.
----------------------------------

    Resolution: Not A Bug

No worries. Hope you got the other module fixed :-)

> UDP sockets used for RTP are sometimes leaked
> ---------------------------------------------
>
>                 Key: ASTERISK-24239
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24239
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.9.0, 11.11.0
>         Environment: Debian wheezy (7.6) 32-bits
>            Reporter: Etienne Lessard
>            Assignee: Matt Jordan
>
> I found on 2 different asterisk (11.9.0 and 11.11.0 respectively) that UDP sockets bound in the RTP range were left open, and that the res_rtp_asterisk module had a high use count.
> On the first asterisk process, there was ~1625 UDP sockets bound in the RTP range and the res_rtp_asterisk use count was 811. On the second, there was 12 UDP sockets bound in the RTP range, and the res_rtp_asterisk use count was 6.
> In both case, there was no active channels (core show channels returned nothing) and there was no recent activity (i.e. no new calls since at least 5 minutes).
> Excerpt of lsof output for the asterisk process:
> {noformat}
> udp        0      0 192.168.17.81:18699     0.0.0.0:*
> udp        0      0 192.168.17.81:11531     0.0.0.0:*
> udp        0      0 192.168.17.81:19211     0.0.0.0:*
> udp        0      0 192.168.17.81:15627     0.0.0.0:*
> ....
> {noformat}
> I've tried to reproduce the problem but I've not yet been able. I've seen that at least one other person had a similar problem: http://forums.asterisk.org/viewtopic.php?f=1&t=82000
> I found this problem while I was looking for a sound problem. On the machine with 811 "leaked" UDP sockets, they had the week earlier a sound problem where A call B, B answer, A could hear B but B could NOT hear A. When that problem happened, an asterisk restart was needed for the calls to once again worked reliably. I did not found any particular errors/warnings in the logs just before the restart.
> I'll update the ticket if I find a way on how to systematically reproduce it.



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