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mrmdev at outlook.com
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 Posted: Wed Apr 27, 2016 8:56 am Post subject: [asterisk-users] SIP/SDP for MulticastRTP page |
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Hi everyone,
I am sending out a multicast page using the following in my dialplan:
| Quote: | | 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 |
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jcolp at digium.com
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| Posted: Wed Apr 27, 2016 8:59 am Post subject: [asterisk-users] SIP/SDP for MulticastRTP page |
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Matthew Murphy wrote:
| Quote: | 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 |
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mrmdev at outlook.com
Guest
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| Posted: Wed Apr 27, 2016 9:51 am Post subject: [asterisk-users] SIP/SDP for MulticastRTP page |
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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.
--Matt
________________________________________
From: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com> on behalf of Joshua Colp <jcolp@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:
| Quote: | 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
--
_____________________________________________________________________
-- 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 |
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jcolp at digium.com
Guest
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| Posted: Wed Apr 27, 2016 9:59 am Post subject: [asterisk-users] SIP/SDP for MulticastRTP page |
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Matthew Murphy wrote:
| Quote: | 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.
| Quote: |
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
--
_____________________________________________________________________
-- 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 |
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