[asterisk-bugs] [JIRA] (ASTERISK-21246) After RTP is negotiated (ALAW) between two SIP endpoints; Asterisk starts sending ULAW marker packets to both endpoints

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21246?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205724#comment-205724 ] 

Rusty Newton commented on ASTERISK-21246:
-----------------------------------------

I looked through the PCAP again along with the codecchange.3 debug. A few observations

* Asterisk settles on the ALAW codec for both call legs
* About ~20 seconds after the ALAW streams start; for no obvious reason Asterisk starts sending ULAW marker packets to both endpoints involved in the call.
* Asterisk is sending the ULAW packets to both end points, but the endpoints are only talking ALAW.
* The ULAW packets are being sent from and to the same ports used in the ALAW stream
* There was no re-negotiation of Codecs. The codec did not change.
* Asterisk starts re-inviting the caller, but SDP is same on both sides so no changes happen and Asterisk ignores the SDP.
* I don't know whats going on..

Are you positive this is from your vanilla, unmodified Asterisk install?

Please provide the rtp.conf, sip.conf and any other relevant configuration files for the peers/users involved.

Please provide a new debug log and pcap, but also enable RTP debug for the Asterisk log. (Sorry I didn't have you do that originally)
                
> After  RTP is negotiated (ALAW) between two SIP endpoints; Asterisk starts sending ULAW marker packets to both endpoints 
> -------------------------------------------------------------------------------------------------------------------------
>
>                 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, codecchange.0, codecchange.2, codecchange.3, 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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