[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=220082#comment-220082 ] 

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

The issue is still 100% reproducible on my end, and I have .  Here's the setup that demonstrates the problem, although we've found other places where this happens:

Call from Asterisk 1 to Asterisk 2, with two SIP phones - 226 and  connected behind NAT.

This is on Asterisk 2, in the incoming SIP context:

[2014-06-27 22:01:18] VERBOSE[1444][C-000001c0] pbx.c:     -- Executing [303xxxxxxx at outside-in:57] GosubIf("SIP/Asterisk-1-00000430", "1?set-jitterbuffer,s,1") in new stack
[2014-06-27 22:01:18] VERBOSE[1444][C-000001c0] pbx.c:     -- Executing [s at set-jitterbuffer:1] NoOp("SIP/Asterisk-1-00000430", "Set Jitter Buffer on Receive Channel") in new stack
[2014-06-27 22:01:18] VERBOSE[1444][C-000001c0] pbx.c:     -- Executing [s at set-jitterbuffer:2] Set("SIP/Asterisk-100000430", "JITTERBUFFER(fixed)=200") in new stack

We then process the call and hand it to our first SIP peer:

[2014-06-27 22:01:18] VERBOSE[1444][C-000001c0] pbx.c:     -- Executing [303xxxxxxx at did-dial:69] Dial("SIP/Asterisk-1-00000430", "SIP/226,20,U(set-did-hint^303xxxxxxx,INUSE)xt") in new stack

Do an attended transfer, from 225 -> 525 (note that we do not re-enable the jitter buffer in this leg of the call):

Audio works normally when 225 and 525 are talking before the attended transfer is completed.  When you finalize the transfer, however, you have one way audio.

Upon completion of the transfer and you are left with a situation where the original call between Asterisk 1 and Asterisk 2 is still pinned up.  RTP is properly flowing between those two boxes, verified by a TCP dump.  However, Asterisk 2 simply eats the RTP coming from the far end and never sends it back to extension 525, thus resulting in the one way audio issue.

In my setup, this is 100% reproducible, and removing the jitter buffer resolves it completely every time.  Hope that helps a little bit.

> 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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