[asterisk-bugs] [JIRA] (ASTERISK-26395) IAX2 doesn't support SILK

Matt Jordan (JIRA) noreply at issues.asterisk.org
Tue Sep 20 16:59:01 CDT 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26395?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232366#comment-232366 ] 

Matt Jordan commented on ASTERISK-26395:
----------------------------------------

You're correct, it doesn't support SILK.

That's even noted in the [IAX2 RFC, section 1.2|https://tools.ietf.org/html/rfc5456#section-1.2]:

{quote}
   While IAX is very effective, addressing many of today's
   communications needs, it does have a few limitations.  For instance,
   IAX uses a point-to-point codec negotiation mechanism that limits
   extensibility because every IAX node in a call path must support
   every used codec to some degree.  In addition, the codec definition
   is controlled by an internally defined 32-bit mask, so the codecs
   must be defined in the protocol, and the maximum number of
   simultaneous codecs is, therefore, limited.
{quote}

If you look at [Section 8.7|https://tools.ietf.org/html/rfc5456#section-8.7], you'll see what formats IAX supports. SILK isn't one of them.

At this time, there are no plans to update the RFC, nor are there any plans to add any media formats beyond what IAX supports today.

Closing out as "Won't Fix" (plus, technically, this is an improvement or feature request, and not a bug).


> IAX2 doesn't support SILK
> -------------------------
>
>                 Key: ASTERISK-26395
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26395
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_iax2
>    Affects Versions: 11.23.1
>            Reporter: Yury Tsaregorodtsev
>
> Hello,
> IAX2 doesn't support SILK
> *CLI> iax2 show peer PEER1
>   * Name       : PEER1
>   Description  : 
>   Secret       : <Not set>
>   Context      : iq_in
>   Parking lot  : 
>   Mailbox      : 
>   Dynamic      : No
>   Callnum limit: 0
>   Calltoken req: No
>   Trunk        : No
>   Encryption   : No
>   Callerid     : "" <>
>   Expire       : -1
>   ACL          : No
>   Addr->IP     : 88.8.1.1 Port 4569
>   Defaddr->IP  : 0.0.0.0 Port 0
>   Username     : me
>   Codecs       : (nothing)
>   Codec Order  : (silk24)
> On Dial remote peer just get HANGUP message from origination asterisk.



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