[asterisk-bugs] [JIRA] (ASTERISK-24543) Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs

[Alexander Traud (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24543?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Alexander Traud updated ASTERISK-24543:
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    Attachment: chan_sip_codec_clash_revision_2.patch

Long story short: I created a regression because I used the wrong variable.

There is one variable and one flag to detect an outgoing call. The variable is used for the session and the flag for the dialog. I should have used the one for the session, because the state of the variable remains. When a Session Timer kicks in the state of the flag is flipped (after the codecs are added to the SDP). Therefore, on the second elapsed Session Timer, Asterisk is sending all allowed codecs instead of the requested one. Or stated differently: The issue was just moved.

Some VoIP/SIP clients (for example Nokia Symbian/S60) interpret this change of the codec list as a changed session and increase the SDP counter. Because of that, Asterisk is creating a new dialog and ignoring the NAT behaviour for the RTP audio. Asterisk is sending the RTP to a local IP address rather than the remote resource. Therefore, after the second Session Timer fired, Nokia Symbian/S60 phones did not receive any RTP packets anymore. A no-audio scenario. The attached patch fixes this.

If anyone is interested, I am able to provide SIP/SDP traces. However, the change/fix is so minor, let us try without any SIP/SDP traces.

> Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
> --------------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24543
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24543
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling, Channels/chan_sip/General
>    Affects Versions: 13.0.0, 13.6.0
>            Reporter: Taylor Hawkes
>      Target Release: 13.7.0
>
>         Attachments: Asterisk-13-SDP.pcap, chan_sip_codec_clash.patch, chan_sip_codec_clash_revision_2.patch, sip.conf
>
>
> Asterisk 13.0 responds to INVITE with SDP of all possible codecs it has available.  
> Asterisk 12.7 responds only with the codecs sent in INVITE request and  whatever it has available (i believe this is correct behavior). 



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