[Freeswitch-users] FS as Media Gateway Only
Saeed Ahmed
saeedahmad1981 at gmail.com
Sun May 30 02:36:37 PDT 2010
Another point, with following script:
media_server="my_media_X.mydomain.com <http://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);
we can bill the customer (inbound) but supplier (outbound) is behind
media_server, so i think it would be a problem to manage ,supplier side for
billing and statistics, on signalling server.
On Sat, May 29, 2010 at 10:55 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
> In our implementation only Media servers generate CDRs. No Media session -
> no CDR (at least we need exactly this behavior).
>
> However if you need, you can generate CDRs only on Signalling servers I
> beleive.
>
> Regards,
> Vitalie
>
> 2010/5/29 Saeed Ahmed <saeedahmad1981 at gmail.com>
>
> 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
>>>
>>> Service Client IPeva
>>> tel: 0811 46 26 26
>>> www.ipeva.fr - www.ipeva-studio.com
>>>
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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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