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

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204213#comment-204213 ] 

Michael L. Young commented on ASTERISK-21246:
---------------------------------------------

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


                
> 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
>         Attachments: negiot-switch
>
>
> 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.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira