[Freeswitch-users] FS as Media Gateway Only
Saeed Ahmed
saeedahmad1981 at gmail.com
Sat May 29 13:44:25 PDT 2010
If i understood correctly, Vitalie solutions is still workable, (although
what Code mentioned, would be ideal), because from customer side its normal
to provide multiple IPs or in most cases a whole subnet range, and call can
come any IP from theatrange, a good example is Arbinet.
Commercial SBCs like nextone support it.
Vitalie, i've a concern that in your solution how would we deal with cdrs?
Thanks
On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com>wrote:
> Code,
>
> you're totally right.
> In this model (FS), the media server will also be in the SIP Path.
> That's why I answered in the first place that this was not achievable with
> FS, because your idea was more a Kamaillo/RTPProxy setup, where the
> mediaserver only does RTP with the endpoints, and is not in the SIP path at
> all:
>
> inbound <--------SIP------ SIP Server/Proxy ------------SIP-------> Carrier
> |
> <---------RTP------ MediaServer--------RTP--------------->
>
>
> Verizon Business (in Europe at least) has a such infrastrucutre: OpenSER
> for the SIP part, and Nortel GWs for the RTP.
> This way, they just give me the IPs of their OpenSER servers, and they can
> deploy as many media servers as they need without telling us (of course, we
> dont filter that).
>
> I don't know how this is implemented in Kamaillo/OpenSER but perhaps, for a
> nice bounty, that would be something possible in FS.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
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>
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> Le 28/05/2010 à 05:34, Code Ghar a écrit :
>
> Hi Vitalie
>
> Thanks for providing the link and details. If I understood correctly, the
> chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
> names and terms in my original question), while the chain of media would be
> Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
> media and minimal servers handling signaling.
>
> Let me clarify a little more my motivation for asking this question in the
> first place. I work with telecom carriers on a daily basis and have seen
> many different architectures. The first biggest problem is how to load
> balance SIP traffic when you are receiving calls, if one server is not
> enough. The second biggest problem is handling all RTP, including
> transcoding. With this architecture, one or two IPs for signaling can be
> handled by most carriers. So you can beef up your hardware for signaling and
> depend less on your carrier's ability to load balance traffic for you. If
> they can do round-robin or failover for two IPs, you are golden. And then
> you can deploy multiple nodes to handle all RTP duties, without having to
> concern your carrier about multiple servers and IPs. But there's one thing
> still missing. Your outbound carrier still needs to allow traffic from
> multiple IPs because now they are dealing with FSRTP instead of FSSIP.
>
> Is there such a solution possible using FS that all signaling, for both
> inbound and outbound carriers, can be handled by a couple of FSSIP nodes
> (depending on the amount of traffic, maybe a few more) while delegating
> media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
> always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
> 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
> seen concern themselves only with which SIP IPs they should allow and don't
> care how many or which media IPs they have to deal with. Any ideas on
> minimizing signaling IPs while adding more media IPs as required?
>
> I have seen re-invite being used in production where you can just let your
> inbound and outbound handle media between them on their own without it going
> through your network. But there are circumstances where people might need to
> pass media through their own network, chiefly to perform transcoding and
> also to handle other interop issues. It is because of this use case,
> combined with the need for minimizing signaling IPs, that the original
> question was asked.
>
>
>
>
> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> 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?
>>>
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>>
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