[asterisk-users] SIP/SDP for MulticastRTP page

Joshua Colp jcolp at digium.com
Wed Apr 27 08:58:25 CDT 2016

Matthew Murphy wrote:
> Hi everyone,
> I am sending out a multicast page using the following in my dialplan:
> Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q)
> Everything works great, but I had a question about SIP and SDP:
> Should I be seeing a SIP/SDP message from the asterisk server containing
> media information and the multicast IP address? On wireshark, I see
> SIP/SDP from the admin phone I am using to dial the extension and
> initiate the page. But I never see a SIP/SDP message with the multicast
> address sent from the Asterisk server to the endpoints. Maybe I
> misunderstand how SIP and SDP fit into the messaging scheme.

You won't. It's up to the phones to be configured to always listen to 
the multicast address and play it out over the speakerphone. This 
eliminates the need to set up a SIP session for each device to have them 
listen in, which can be problematic.


Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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