[asterisk-dev] possibly dumb IAX question

Ast Exp astexp at gmail.com
Mon Apr 2 16:35:00 MST 2007


On 4/2/07, Tim Panton <tim at mexuar.com> wrote:
>
> On 2 Apr 2007, at 06:40, Ast Exp wrote:
>
> > I have what may be a dumb IAX (and possibly SIP) question here, but I
> > wanted a clear answer
> >
> >
> > What would happen if Asterisk received a SIP connection with two media
> > streams, and that needed to be connected to an IAX (or to a lesser
> > extent a SIP) client registered with Asterisk? The case I am thinking
> > of is the first media stream being either voice or video (h.264), and
> > the second stream being some unidentified binary protocol used to send
> > mechanism control data to a remote control device.
> >
> > It wasn't clear to me how the IAX protocol handles two concurrent
> > media streams to the same endpoint.
>
> If you mean 2 streams within the same Call...
> It doesn't. IAX assumes that the media stream contains the
> necessary data multiplexed into one stream at the codec level.
>
> However if you want to send 'mechanism control data' there are a
> number of messages that might be appropriate depending on the
> nature of the data.
>
> There was a discussion a while back about how best to extend IAX
> to carry arbitrary call related status and command data, the conclusion
> of which was that for the moment you either have to wrap it up in ascii
> and send it in a Text frame or extend the protocol with a new message
> type
> to cover it.
>
> Tim.

So if I understand you correctly, if an inbound SIP call tries to
negotiate an alternate and separate secondary media stream, IAX will
not be aware of the secondary media stream because it is not
monolithically multiplexed? Does this mean Asterisk will simply drop
the secondary media stream, or will the call fail?

And as a double check, will Asterisk properly forward secondary media
streams to registered SIP clients? The idiot check here being, if the
device mechanism control data is unknown (not a built in Asterisk
codec), but both SIP endpoints speak the same "codec" (SIP client is
registered with the unknown "codec" name), will Asterisk perform pass
through?


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