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

[Peter Katzmann (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205197#comment-205197 ] 

Peter Katzmann edited comment on ASTERISK-21246 at 4/11/13 3:24 AM:
--------------------------------------------------------------------

Hello,
two new files uploaded. CPAP Trace and corrsponding debug log as suggested
this trace has the branch/1.8 svn asterisk as base

peter
                
      was (Author: pk16208):
    Hello,
two new files uploaded. CPAP Trace and corrsponding debug log as suggested

peter
                  
> 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, codecchange.0, codecchange-filtered.pcap.gz, filtered1512-2-leg.pcap, filtered1512-2.pcap, 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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