SIP/SDP For MulticastRTP Page

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Asterisk Users 3 Comments

Hi everyone,

I am sending out a multicast page using the following in my dialplan:


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.



3 thoughts on - SIP/SDP For MulticastRTP Page

  • Matthew Murphy wrote:

    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.


  • 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.


  • Matthew Murphy wrote:

    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.

    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.