[asterisk-users] SIP/SDP for MulticastRTP page

Joshua Colp jcolp at digium.com
Wed Apr 27 09:59:39 CDT 2016

Matthew Murphy wrote:
> 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.

Normally you'd have to create an individual channel for each outgoing 
leg, so given 100 phones that's 100 channels you have to create all at 
once. Even using the same multicast stream that's a lot of stuff to 
happen all at once. It also takes some time, so phones at the end may 
find the audio slightly cut off unless you synchronized everything so 
the pager wasn't allowed to provide audio until everyone has confirmed - 
which is more more to do.

As for SDP it's just a way to convey the details for media streams and 
codecs. As well as negotiate it of course.

> 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?

There's nothing built in to do it... so you'd have to do something. 
Other phones just allow it to be provisioned. If you did it outside of 
channel creation it's still lots of work to coordinate things.

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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