[asterisk-bugs] [JIRA] (ASTERISK-18618) IAX2 needs to be converted to use bitmask lists for codec selections

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-18618?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225117#comment-225117 ] 

Matt Jordan commented on ASTERISK-18618:
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This actually can't be solved.

The IAX2 protocol defines in its RFC the allowed media formats:

https://tools.ietf.org/html/rfc5456#section-8.7

SILK is not included. As such, any IAX2 session that is attempted with a media format capability not listed in that section will fail. Changing that would require a modification to the RFC, which is highly unlikely to ever occur.

> IAX2 needs to be converted to use bitmask lists for codec selections
> --------------------------------------------------------------------
>
>                 Key: ASTERISK-18618
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18618
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_iax2
>    Affects Versions: 11, 10.0.0-beta1
>         Environment: Centos 5.7
>            Reporter: Nikos Patronas
>      Target Release: 11
>
>         Attachments: debug_error_iax_silk.log, debug_success_additional_codecs.log, debug_success_sip_silk.log, extensions.conf, iax.conf, sip.conf
>
>
> SipPhone --sip/alaw--> Ast10B1 --sip/silk--> Ast10B1 --...-- ===> SUCCESS
> SipPhone --sip/alaw--> Ast10B1 --iax2/silk--> Ast10B1 --...-- ===> ERROR
> When trying to trunk a call from a sip device to a 10.0.0 Beta1 server to another same server via sip, I can force to use silk (8,12,..) between the servers entering disallow=all \ allow=silk8 in the trunk context in sip.conf and the call is placed successfully.
> When trying the same thing using iax2 channel between servers, the first server claims "Don't know any of (nothing) formats" and "Unable to create translator path for (nothing) to (alaw)" before hanging up. No debug output here because no iax call is attempted.
> As soon as I add eg. alaw to the allow parameter (allow=silk8,silk12,alaw) the connection establishes using alaw of course. Then the debugger indicates
> CODEC_PREFS     : (alaw|silk8|silk12|ulaw)
> FORMAT          : 8
> FORMAT2         : alaw
> CAPABILITY      : 8
> CAPABILITY2     : alaw
> (yes, without declaring ulaw!)
> I attach the configuration and debugger output files. In case I change the "iax2" word in "Dial" function (extensions.conf) with "sip" the call is placed normally using silk through sip



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