[asterisk-users] SIP/SDP for MulticastRTP page

Matthew Murphy mrmdev at outlook.com
Wed Apr 27 08:55:30 CDT 2016


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.


Can anyone tell me if I should see SIP/SDP coming from my Asterisk server to my endpoints? I hope my question makes sense.


Thanks,


--Matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160427/c555428b/attachment.html>


More information about the asterisk-users mailing list