October 25, 2005

VoIP Peering

Yesterday Neustar announced a new set of “application peering services” called SIP-IX. According to Light Reading, SIP-IX is not just ENUM services. It seems, “SIP application calls, or “sessions” are a little more high-maintenance.” For a SIP session to traverse from one network to another, the two networks must “negotiate a number of other SIP-oriented parameters like provisioning and OSS, SIP security, QOS, and traffic accounting and settlement.” This is according to Mark Foster, CTO of NeuStar. Dan Dearing, Vice President of Marketing, NexTone expresses similar points in an interview with Tehrani. Tehrani has also predicted that 2006 will be the year of VoIP peering. Irwin Lazar comments on these and points out that there is a mailing list discussing the issues of peering.

I am all lost. I have a rather simplistic view of the issues associated with peering. Peering is needed for two reasons – to share the directories and to carry the media traffic across the network boundary.

The former makes sense because the carriers hoard the identity analogous to email ids. Personally I prefer that IP Communications follows the Domain Name model, in the sense that my id is not determined by the service provider I presently use.

The latter is perplexing. It is not clear how many service providers offer QoS. If a service provider offers unlimited in-network calls, then one can be reasonably certain, they do not offer QoS. But if they do offer, then haven’t we recreated PSTN model of interconnecting? Richard Stastny asks the same question.

In any event, for the end users, all these activities are inconsequential. A service that participates in a SIP-based peering program might use a proprietary scheme towards the end-user (as Skype is doing it). Even if a service provider uses SIP nominally, they can make it a close system (as Vonage has done). For the end-user, a major advantage of a SIP based system is that the client can be sophisticated and exciting features will be made available to them. These are not guaranteed just because SIP is used for peering. Just look at PSTN. It uses SS7 and ISUP for interconnection, but the access could be Tip and Ring and the phone could be a rotary phone.

Posted by aswath at October 25, 2005 05:32 PM
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Comments

Aswath, if we go to a domain name model I assume you mean that all SIP URIs are resolvable, and anyone can call anyone else provided they know the called party's URI?

If so, then how we stop SPIT? Will folks want a model that requires them to set up white lists?

Posted by: Irwin Lazar at October 25, 2005 10:06 PM

Irwin, I don't mean that it should be universally resolvavble. The main objective is that the URI be portable. To this end, the universal URI could be mapped to a restrictive one and can do whatever is needed to minimize SPIT.

Posted by: Aswath at October 26, 2005 12:50 AM



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