[asterisk-users] Sip and the media path

[qasimakhan at gmail.com]

Hi David,

Direct media should work either way. if your phones are behind NAT you will
also require the NAT option enabled in asterisk, How ever  the tricky part
in all this is that you wont be able to acurately keep track of calls on
these phones. If or any unforeseen reason the phone goes offline or you
dont recieve BYE signal, asterisk wont be able to know that the call has
ended. So if call information is critical for you then byepassmedia is not
recomended for you.

Regards,
Qasim


On Thu, Apr 25, 2013 at 8:48 PM, David Wessell <david at ringfree.biz> wrote:

>  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
> --
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