[asterisk-bugs] [JIRA] (ASTERISK-21246) Codec change in rtp stream during call (alaw to ulaw)
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Wed Apr 17 17:39:01 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205390#comment-205390 ]
Rusty Newton commented on ASTERISK-21246:
-----------------------------------------
Peter,
The calls in the pcap don't seem to appear in codecchange.0 log file. The calls in the pcap have call-id *3c2683f54ccd-awzivjl30opn* and *2eede5053b53a84e065150367d4ececa at 10.0.0.5:5060*. I don't see these in the log file.
> 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: 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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