[asterisk-bugs] [JIRA] (ASTERISK-24747) Call is not terminated in time due to lack of RTP nor it is shutdown if peer sends BYE

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24747?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225267#comment-225267 ] 

Rusty Newton commented on ASTERISK-24747:
-----------------------------------------

{quote}
but we hope that with the sip debug on and the debug level enabled we can provide you with the information that you need. 
{quote}

Did you attach the new log with 'DEBUG' type log messages (please also, level 5 or above)? I don't see it.

I don't have a specific opinion on your options for a work-around. Whichever works for you is fine for a work-around. If you can get the new log up I'll take a look at that and get a developer to review it if it has anything useful.

> Call is not terminated in time due to lack of RTP nor it is shutdown if peer sends BYE
> --------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24747
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24747
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.2.1
>         Environment: Linux 2.6.18-274.7.1.el5 #1 SMP Thu Oct 20 16:21:01 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux
> CentOS release 5.7 (Final)
>            Reporter: Fabian Borot
>            Assignee: Rusty Newton
>         Attachments: asterisk log.txt
>
>
> We use RTP time out as a protection to shutdown the calls when there is lack of RTP for 30 secs. We have been noticing lately this behavior, either the calling or called peer terminates the call and asterisk takes a long time, (variable, around 30-40 secs later) to forward the BYE to the other leg.
> At the same time, asterisk shows in the logs for those calls that the call will be terminated due to Lack of RTP activity in 30 secs, we see the message "Rescheduling destruction for 10000 ms" for the channel, several times, not even spaced out every 10000 ms but 40 secs, 50 secs etc until eventually (sometimes after 4 mins) the call is terminated, causing to log the duration of the call with wrong values. Our customer are complaining  about it. We restarted the asterisk and so far it hasn't happened again but it looks like this behavior lasted several weeks and we don't know yet the repercussion of this issue because it may take a couple of billing cycles for the customers to create the disputes.
> The weird thing is that the BYE sent by the peers is acknowledged with the 200 OK but the calls are not terminated, they are terminated after several cycles of that "Rescheduling destruction" msgs.
> Looks like while the asterisk is in the process of terminating the call due t lack of rtp that the BYE msg is not taking into account to terminate the call. Is this possible ?
> thank you
> Any ideas?
>  Has anybody experienced an issue like this before?



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