[asterisk-bugs] [JIRA] (ASTERISK-23812) One Way Audio on REFER with Jitterbuffer On

[JoshE (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23812?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=219212#comment-219212 ] 

JoshE commented on ASTERISK-23812:
----------------------------------

This does occur on all Asterisk 11 versions I am aware of.  I had just done most of the testing on versions between 11.3 and 11.9.

I can pull a PCAP on this, but the scenario is pretty straightforward and is 100% reproducible.  The net of it is that with the JB enabled, after transfer, Asterisk does not send RTP at all to the recipient of the transfer request.

> One Way Audio on REFER with Jitterbuffer On
> -------------------------------------------
>
>                 Key: ASTERISK-23812
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23812
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.3.0, 11.9.0
>            Reporter: JoshE
>            Assignee: Rusty Newton
>            Severity: Critical
>
> We are seeing one way audio on transferred calls with the jitterbuffer running.  This affects ALL versions of Asterisk 11 from 11.3 to 11.9.
> Inbound external call treated with:
> Set(JITTERBUFFER(fixed)=200)
> Answered by a SIP extension on Asterisk.  This is transferred to another SIP extension on the same PBX.  When the transfer is completed, and this can be either attended or blind, you will have one way audio.  The internal party's audio makes it out to the original inbound call, but the person to whom the call was tranferred cannot hear.
> Turning off the jitterbuffer 100% resolves the issue.  This could possibly be related to ASTERISK-21144.



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