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

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-21246:
------------------------------------

    Status: Waiting for Feedback  (was: Triage)

Peter we'll probably need an Asterisk log file with SIP debug turned on as well. That log should correlate with the traffic in your two latest pcaps:

filtered1512-2.pcap

filtered1512-2-leg.pcap

If you can't get that now, then you'll need to provide new pcaps with new Asterisk log files.

Essentially we need the pcaps to show all the SIP and RTP traffic. Then there needs to be corresponding Asterisk log files that contain SIP debug output along with the VERBOSE and DEBUG message types (both turned up to level 5).

Sorry for any confusion!





                
> 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, 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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