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

David Woolley (JIRA) noreply at issues.asterisk.org
Wed Apr 3 07:53:01 CDT 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204907#comment-204907 ] 

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

The pcap seems to be missing the most important leg and the debug is missing sip debug output.

However, my guess is that Asterisk has native bridged the call (packet to packet), because the negotiated codecs are compatible. 10.107.160.11 has then either unilaterally changed the codec, or has tried to re-invite to it (i.e. the 415 is the response to that re-invite, which somehow hasn't got logged), but had the re-invite rejected, then changed the codec anyway.

Because Asterisk is native bridging, it is trusting 10.107.160.11 to only use agreed codecs, so doesn't notice the codec change and therefore does not drop out of the native bridge and start transcoding.

A work around may be to enable a feature that is incopatible with native bridging (because it needs the media stream to be inspected).

An accurate and complete capture of the leg to 10.107.160.11 is really needed to confirm that it is at fault.
                
> 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.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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