[asterisk-bugs] [JIRA] (ASTERISK-21246) Codec change in rtp stream during call (alaw to ulaw)

[David Woolley (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204431#comment-204431 ] 

David Woolley commented on ASTERISK-21246:
------------------------------------------

A midstream change that can confuse some phones is if there is a discontinuity in the timestamps, but no corresponding change in SSRC.  At least as of early 1.6, Asterisk only set the marker bit in this case, because there was no mechanism to propagate the SSRC between incoming and outgoing channels, when non-natively bridged.  The marker bit is not reliably transmitted, and its semantics are an oiptional opportunity to flush latency buffers, not the invalidation of the time stamp.

We only found this problem on Cisco phones, and worked round it by incremetning SSRC whenever the marker bit was set.
                
> Codec change in rtp stream during call (alaw to ulaw)
> -----------------------------------------------------
>
>                 Key: ASTERISK-21246
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21246
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Resources/res_rtp_asterisk
>    Affects Versions: 1.8.17.0, 1.8.18.0, 1.8.19.0, 1.8.20.1
>         Environment: ubuntu 10.04
>            Reporter: Peter Katzmann
>            Assignee: Peter Katzmann
>         Attachments: CallWithTranscoding, myDebugLog, myDebugLog, negiot-switch, vanilla-trace.txt, ws-trace.txt
>
>
> See attached traces
> During call asterisk switches the negotiated codec (alaw) in the rtp to ulaw.
> If there is no renegotiation (like on my snom 720 sometimes), the user has a audible distortion.

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