[asterisk-bugs] [JIRA] (ASTERISK-21226) one way audio after call was on hold

[Stefan Schmidt (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-21226?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Stefan Schmidt updated ASTERISK-21226:
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    Attachment: Holdbug-codecs.txt

in this txt you can see the codec states during the call, during the call is placed on hold, and after it. After the call is put on hold, asterisk thinks its ok to go into P2P bridging cause both sides uses the same codec even if its not right.
                
> one way audio after call was on hold
> ------------------------------------
>
>                 Key: ASTERISK-21226
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21226
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_ulaw
>    Affects Versions: 11.2.0
>         Environment: debian 6 in an openvz virtual machine
>            Reporter: Stefan Schmidt
>         Attachments: Holdbug-codecs.txt, moh-onewayaudio-NOK, moh-onewayaudio-OK
>
>
> I have a sip client which offers ulaw and alaw in this order. my system allowes both but prefer alaw. When i have an incoming call using alaw, going through asterisk dialing this client the sdp offer is alaw and ulaw. The client replies with a normal 200 ok with only ulaw. This call leg to the client is then running on ulaw and everything is fine. When the client puts the call on hold and take it back (a=sendonly & sendrecv) the call then changes to alaw and the client audio to * is not recognized at asterisk anymore.
> In 1.8 this behavior worked without any problems but since 10 one way audio happens.
> I can see in debug log that asterisk sets the write format to alaw but the marker bit isnt recognized. The difference to 1.8 is that asterisk didnt back out of the bridge cause the format has changed.

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