[OpenSIPS-Users] OpenSip SIP, SIP-I e SIP-T

[Adrian Georgescu]

Hm,

It is very hard to judge the benefits of performing all the nice to  
have feature at a higher level protocol while still having to support  
legacy expensive infrastructure underneath.

Now, last time I heard about SIP-T was by an ECMA standard a few years  
ago. ECMA is a sort of inverse pyramid European standards body that  
nobody listens to. Basically, they are sponsored by vendors to endorse  
'standards' because they posses an EU stamp. The word here in Europe  
goes that if something went to the extent of geting an ECMA official  
endorsement, one knows that it is a standard with no future and no  
company invests in it anymore.

Maybe I am wrong and this has much more sense in the US.

Adrian


On Feb 19, 2009, at 8:43 PM, Alex Balashov wrote:

> To expand on this just a little bit:
>
> While here in the VoIP cottage industry we mostly deal with SIP to  
> begin with, in that we use ISDN gateways for connecting to carriers,  
> get SIP trunking from our carriers/ITSPs, and so on, the reality is  
> that most stuff in the PSTN carrier space is still done with big- 
> iron TDM equipment, at least here in the US.  If you want to be a  
> competitive carrier, you *must* interconnect with the incumbent  
> telco using SS7;  no ands, buts, ors.
>
> That doesn't mean there aren't a lot of opportunities to deploy SIP  
> internally inside the service delivery core.  The main benefit SIP  
> provides there is that it is so high-level and easy to manipulate.   
> As a result, a lot of mediation, logging, billing, analysis,  
> translation, LCR  can be done easily and inexpensively.  Before SIP  
> and H.323 came along, doing this kind of stuff required a box that  
> did all that and spoke SS7 or, at the very least ISDN Q.931, and  
> that is much more expensive, inflexible, and difficult to manipulate.
>
> Promoting this traffic to a higher-level protocol stack that has  
> more applications and tools to deal with it allows the development  
> of solutions for doing sophisticated telco-world stuff using  
> commodity hardware and open methodologies, open-source style.  That  
> has triggered a wave of new products and paradigms in the telco  
> space in a way that is analogous to how Asterisk et al have  
> revolutionised the PBX space.
>
> One example of this is TransNexus' NexOSS/NexSRS product (www.transnexus.com 
> ).  They use the OSP (Open Settlement Protocol) module for OpenSER  
> and/or for Asterisk (depending on whether a B2BUA is required)  
> internally inside their product to perform a lot of neat AAA and  
> routing functions (e.g. the NexSRS route server).  Their ability to  
> do this benefits precisely from the fact that the traffic can be  
> moved onto a higher-level protocol plane and away from proprietary,  
> expensive, closed and inflexible stuff that has been a defining  
> feature of the telco world.  If you can turn the traffic into SIP or  
> H.323, they can deal with it, but if it's SS7 or PRI, they can't.   
> The world is going more "soft[ware]."
>
> At the same time, the telco space is not a SIP world right now;  the  
> network edges are still SS7, and the market really hasn't settled on  
> a good private SIP interconnection/peering strategy and  
> implementation for intercarrier settlement. So, for the most part  
> SIP trunking is used for customer access only.  The SS7 information  
> must be conserved in this type of setup, and that's one of the  
> reasons the sort of thing that SIP-T is exists.
>
> Alex Balashov wrote:
>
>> Adrian Georgescu wrote:
>>> Why should SIP-T still exist? Is it cheaper than having a gateway?  
>>> What is the practical use case for investing in such technology?
>>>
>>> I am eager to learn
>> We've used it extensively in work with CLECs that operate TDM  
>> switches such as the Metaswitch, Lucent LCS/Telica, etc.
>> When a carrier operates more than one switch, SS7 interconnection  
>> between them is generally required so, for the same basic reasons  
>> an internal iBGP mesh or partial mesh (confederation) between two  
>> border routers is required for IP.   One switch must be aware of  
>> numbers routed or ported into the other switch, and so on.
>> The reason for its existence is that if both network elements  
>> support SIP-T, it allows you to replace an SS7 IMT (inter-machine  
>> trunk) with an IP-based mechanism for this interconnection.  This  
>> allows you to move the traffic over a data network and get all the  
>> benefits that this brings;  economies of scale through decreased  
>> facilities, oversubscription, etc.  The main benefit is the  
>> elimination of TDM trunk exhaust;  SS7 IMTs are physically bundles  
>> (trunk groups/TCICs) of DS0s, usually consisting of one or more  
>> T1s, and sometimes DS3s or more.  That means that when a large  
>> volume of calls is running between the two switches, you could burn  
>> up all your SS7 trunks.  Running the calls as SIP-T allows you to  
>> use something like a gigabit network core to make that problem go  
>> away somewhat -- a key benefit of VoIP in most other scenarios with  
>> which you are familiar with.
>> At the same time, the switches still need ISUP attributes carried  
>> in SS7 IAMs and ACMs for billing, because that's just the  
>> information they operate on internally.  SIP-T provides an IP-based  
>> way to encapsulate that information.
>> SIGTRAN (essentially, SS7-over-IP) is another way to do this.   
>> However, SIP-T is lightweight and easier to deploy.  It also allows  
>> you to use existing SIP network elements (proxies, session border  
>> controllers, etc.) to route and manage the traffic.   For example,  
>> if you were using OpenSIPS + ACC + FreeRADIUS as a CDR catcher, you  
>> could run the "SS7" calls between two switches and log the  
>> appropriate information as custom attributes.  There are no good  
>> open-source implementations for SIGTRAN - nothing as turn-key as  
>> Kamailio or OpenSIPS.  SIP is high-level and much easier to deal  
>> with and manipulate using a far wider range of tools.
>> SIP-T is also becoming an attractive external interconnect option.
>
>
> -- 
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775

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