[asterisk-bugs] [JIRA] (ASTERISK-25446) Warnings with jitter buffer enabled and transcoding from G722 to ulaw

Asterisk Team (JIRA) noreply at issues.asterisk.org
Mon Oct 5 13:27:33 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25446?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227778#comment-227778 ] 

Asterisk Team commented on ASTERISK-25446:
------------------------------------------

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].

> Warnings with jitter buffer enabled and transcoding from G722 to ulaw
> ---------------------------------------------------------------------
>
>                 Key: ASTERISK-25446
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25446
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: 11.19.0
>         Environment: Centos 6.5 x64
>            Reporter: Eli Hunter
>            Severity: Minor
>         Attachments: codecs.conf, sample_sip.conf
>
>
> This is the same as ASTERISK-24424. 
> I'm seeing this issue with Polycom endpoints using G722 with ulaw at the other end after enabling the jitter buffer on Asterisk 11.19.
> WARNING[11984][C-0000076d]: abstract_jb.c:284 ast_jb_put: SIP/18-RST-000019c7 received frame with invalid timing info: has_timing_info=0, len=20, ts=17140, src=slin 8000khz -> 16000khz
> The only options in my rtp.conf are
> rtpstart=10000
> rtpend=20000
> sip.conf and codecs.conf are attached



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