[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=204432#comment-204432 ] 

Peter Katzmann commented on ASTERISK-21246:
-------------------------------------------

Hello,
i can deliver you a pcap file if it will help.
The snom phones with 8.4 and 8.7 firmware series are hte problem in out case. The Siemens OpenStage seems to work reliable.
But you said when Both sides Allow the codec, but only the patton side allows the usage of ulaw, the asterisk side has the following config:
disallow=all
allow=alaw h261 h263 h263p h264, and the phone was configured to answer only with one codec.

                
> 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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