[Freeswitch-users] FS as Media Gateway Only
Vitalii Colosov
vetali100 at gmail.com
Sun May 30 02:45:37 PDT 2010
Could you please provide an example of what exactly do you think is a
problem?
Because now I don't have any problem with this configuration and I am
wondering what is the exact scenario where we cannot use such configuration.
Thank you,
Vitalie
2010/5/30 Saeed Ahmed <saeedahmad1981 at gmail.com>
> 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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