[asterisk-users] SIP/SDP for MulticastRTP page

Matthew Murphy mrmdev at outlook.com
Wed Apr 27 09:50:43 CDT 2016

Thanks Josh,

I have actually built my own endpoints and was experimenting with dynamically creating multicast sessions so that I didn't need to pre-configure the multicast addresses at all. When you say, "...This eliminates the need to set up a SIP session for each device to have them listen in, which can be problematic." What do you mean by "problematic"? I was just curious. I thought SDP was built for this kind of thing, but I don't know the history and I am sure there are things I haven't thought of when it comes to implementation, security, etc.

Also, do you have any thoughts on setting up multicast sessions without a priori knowledge on the endpoints? Would I have to spin my own message protocol to do this? Could I monkey around in the Asterisk source to make it work? Or, is it just a huge waste of time and effort?

I really appreciate your quick response to my earlier question. Thanks a lot for your time.


From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Joshua Colp <jcolp at digium.com>
Sent: Wednesday, April 27, 2016 9:58:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP/SDP for MulticastRTP page

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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