[asterisk-bugs] [JIRA] (ASTERISK-25270) rtptimeout doesn't work at all when using JitterBuffers of any kind

Asterisk Team (JIRA) noreply at issues.asterisk.org
Tue Jul 21 07:26:33 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25270?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226977#comment-226977 ] 

Asterisk Team commented on ASTERISK-25270:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> rtptimeout doesn't work at all when using JitterBuffers of any kind
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-25270
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25270
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/Jitterbuffer, Core/RTP
>    Affects Versions: 13.4.0
>         Environment: Debian 7 VM
>            Reporter: Florian Loyau
>
> Setting rtptimeout in SIP Conf is supposed to terminate the call when no RTP Information was received after a certain time.
> However this doesn't seem to be the case when using any kind of JitterBuffers (either through jbenable and jbimpl in sip.conf as i tested, or according to an old forum post, when using the JITTERBUFFER() DialPlan application).
> Test Procedure:
>  - Enable JitterBuffers in sip.conf
>  - Setup an extension throwing into a StasisApp
>  - Have the StasisApp make the channel join a bridge
>  - Abruptly cut the SIP/RTP client through a SIGKILL or network connectivity loss
> Expected Result:
>  Asterisk detects the lack of RTP traffic and terminates the call after the set timeout, notifying in Console, and the ARI Application via StasisEnd/ChannelLeftBridge/ChannelDestroyed
> Actual Result: 
>  Nothing happens, call goes on despite receiving no data
> Exact same setup with jbenable=no works as expected. 
> Apparently this bug has been around for a couple years, since I noticed a few issues on the bugtrackers and around forums from 11.0.x that could be linked to this one..



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