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

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23812?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=220086#comment-220086 ] 

Matt Jordan commented on ASTERISK-23812:
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A pcap would really be useful here. A SSRC change, timestamp jump, or marker bit may have been set on the inbound RTP stream, and we may not have properly reset the jitter buffer in that case. A pcap would confirm what the root cause of the problem is.

> 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
>          Components: Channels/chan_sip/Transfers
>    Affects Versions: 11.3.0, 11.9.0
>            Reporter: JoshE
>            Assignee: JoshE
>            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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