[Freeswitch-users] FS as Media Gateway Only

Thu May 27 06:51:49 PDT 2010

Also consider using a sip proxy to distribute the calls to the cluster of
media boxes.
This is one of the main reasons we have suffered through SIP so we could
have proxies and redirectors based on http =/

On Thu, May 27, 2010 at 1:42 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Vitali, Code,
>
> I apologize, I answered too quickly.
> That's actually a very smart way to do it when you don't have/want a
> proprietary protocol.
>
> It should be possible to distribute the calls evenly by using mod_limit.
> You can even take the transcoding into account.
> If you know your media servers will only use G711 to the outside, you can
> call mod_limit once if the inbound call is G711 also, but you may call
> mod_limit twice or thrice or more (to be calculated) if the inbound call is
> G729.
> This way, your mod_limit figures per media server will reflect the actual
> load on the server, not only the number of current calls.
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pièces jointes sont confidentiels et établis à
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autorisée est interdite. Tout message électronique est susceptible
> d'altération. **IPeva** décline toute responsabilité au titre de ce
> message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas
> destinataire de ce message, merci de le détruire immédiatement et d'avertir
> l'expéditeur.*
> *
> *
>
>
>
> Le 27/05/2010 à 08:15, Vitalii Colosov a écrit :
>
> Hi Code,
>
> I have working example doing exactly what you've described.
> One signalling FS bridges incoming call to a set of media servers
> (depending on ip, but you can implement any routing logic including round
> robin) and then transfers media stream after bridging to that media server.
>
> You can achieve this on signalling FS by creating a Lua script that
> contains the following lines:
>
> media_server="my_media_X.mydomain.com"; --to be determined by routing
> logic
> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> The call will arrive to the selected media server successfully and media
> stream will start bypassing signalling FS after bridge.
>
> You can read the following thread, it describes how you can setup such
> configuration.
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>
> I think it will fit your needs.
>
> Regards,
> Vitalie
>
>
> 2010/5/27 Code Ghar <codeghar at gmail.com>
>
>> Is it possible -- and are there any case studies, practical experience,
>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
>> request FSRTPx for media resources for each new call and add its IP and port
>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
>> only deals with signaling. This way multiple servers could be deployed to
>> handle media responsibilities and only a handful would be required for
>> signaling. In future if there's a greater need for transcoding, etc. all you
>> need to do is deploy a media server and not have to add servers for
>> signaling.
>>
>> This idea came to me because I have come across two proprietary
>> applications that do it this way. They have a SIP component and a media
>> component. You can run both on the same physical machine or you can separate
>> them out into multiple machines.
>>
>> Another way for this could be to integrate FS as a media component to
>> another application's SIP component. A mix-and-match, so to speak.
>>
>> On the flip side, deploy FS as a SIP server and use media capabilities of
>> some other hardware or software application. For example, FS handles
>> signaling and use dedicated hardware for media. A good example of this is
>> illustrated (somewhat) by an image on Sangoma's website:
>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>> Look at the "pooled transcoding".
>>
>> Is FS even designed to be this modular? If so, how can the aforementioned
>> scenario(s) be achieved?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100527/b1e9cdfb/attachment.html 