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.
Cheers, -- 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
