[asterisk-users] Sip and the media path

[Kevin Larsen]

David,

you obviously have to test for your situation, but the short answer is 
that it should. The connection will start with running through Asterisk, 
but very quickly the phones will see that they can talk directly and take 
the Asterisk server out of the media path. There are a couple of gotchas 
that can happen based on your dial options, so check out this page:
http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite 

canreinvite was renamed to directmedia in Asterisk 1.6.2, but the page is 
still pretty good with regards to the options that are available.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   David Wessell <david at ringfree.biz>
To:     Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users at lists.digium.com>, 
Date:   04/25/2013 10:49 AM
Subject:        Re: [asterisk-users] Sip and the media path
Sent by:        asterisk-users-bounces at lists.digium.com



Kevin,

Thanks for the info. Clarification. The asterisk server is NOT on the same 
LAN as the phones. The asterisk server is in a datacenter only accessible 
via WAN.

However, all of the phones are in side of the same LAN. Will directmedia 
still function that way?

Thanks
David

From: Kevin Larsen <kevin.larsen at pioneerballoon.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users at lists.digium.com>
Date: Thursday, April 25, 2013 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Sip and the media path

You will want to look at the directmedia option. You will want all the 
phones on the same lan as the Asterisk server to be directmedia=yes and 
the ones on the wan to be directmedia=no. Then, internal calls will send 
the media between themselves without involving Asterisk, but ones outside 
on the wan will be forced to talk directly to the Asterisk server for 
everything. You might also want to look at the nonat option of 
directmedia.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 



From:        David Wessell <david at ringfree.biz>
To:        Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users at lists.digium.com>, 
Date:        04/25/2013 07:33 AM
Subject:        [asterisk-users] Sip and the media path
Sent by:        asterisk-users-bounces at lists.digium.com



We're running asterisk 1.8 in the DC on a public IP address.

Connecting to it are about 200 phones behind a LAN in a remote location.

Is there a way to reliably keep asterisk out of the media stream on 
internal calls inside that LAN? All phones are Polycom Soundpoint phones.

Asterisk would say in the media stream for any calls that traverse from 
LAN to WAN. However it would step out for LAN to LAN calls.

Thanks 
David 
--
_____________________________________________________________________
-- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130425/3c3119de/attachment.htm>