asterisk users - Mar 2006 - Multicast Music on Hold

As I understand there is provisions for hold in both RFC 2543  (SIP)  
and RFC 3264 (SDP) and asterisk users the latter.

The RFC 2543 method tells the UA its media stream is at 0.0.0.0 where  
as the method via SDP can tell it to listen to any address/port/ 
protocol combination, which is how asterisk tell it to listen to the  
audio stream it presents when it is asked to hold a call.

Please stop me here if I have totally misunderstood the concept :)

As i am using a central asterisk box with multiple stub sites I don't  
wish every call put on hold to be wasting WAN bandwidth, I am  
wondering if it is possible to create a multicast stream to each site  
and rather than asterisk sending its address and the media  
information during a hold it sends the multicast address and multiple  
phones can be served by the one stream ?

Regards,

Nathan

Nathan Alberti wrote:
> As i am using a central asterisk box with multiple stub sites I don't
> wish every call put on hold to be wasting WAN bandwidth, I am wondering
> if it is possible to create a multicast stream to each site and rather
> than asterisk sending its address and the media information during a
> hold it sends the multicast address and multiple phones can be served by
> the one stream ?
This is certainly possible, but I doubt that most existing SIP phones
have the ability to subscribe to multicast groups and handle it
properly. Without that, it would only work on a LAN (non-routed).