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VoIP Feature Article

VoIP

March 14, 2006

Peering to Presence: Which Will Win First?

Eric Dean, The Heiden Group


Native voice over IP should bypass the circuit-switched PSTN when sending calls between VoIP operators.

Using an IP-based network, VoIP calls route between carriers rather than rely upon TDM infrastructure. Therefore, VoIP peering allows for any VoIP network to terminate calls to another other VoIP carrier. Residential VoIP, enterprise IP telephony and mobile phones all should provide for seamless, native-IP, calling between one another.

How long will it be before all broadband-to-broadband VoIP is free? As soon as the common VoIP infrastructure embraces a scaleable peering model. From a service provider perspective, a SIP trunk to a VoIP carrier technically qualifies as VoIP peering but to a lesser degree. A broadband VoIP operator exchanging calls with Skype (News - Alert) provides a more exciting VoIP peering scenario.

Without VoIP peering, VoIP operators must convert to back to analog TDM and send the calls via the PSTN. While the PSTN provides an extremely reliable, robust communications infrastructure, it also requires significant capital and overhead. Rather that convert from VoIP-to-TDM and back to VoIP, natively exchanging VoIP-to-VoIP calls between carriers provides the most efficient method of inter-communications.

TDM-based voice exchanges and arbitrage companies have attempted to build a community of carriers however most arbitrage companies use classic TDM-based tandem switches charging per minute like any other inter-exchange carrier. This approach still requires the customer to backhaul their voice to a collocated PoP for interconnection. Some carriers now offer VoIP peering but still require both voice signaling and media to transit their soft switches while charging a per-minute fee.

VoIP signaling and media do not need to follow the same path and in most cases will not. If two VoIP companies peer their networks at a collocation facility and a VoIP residential customer calls a customer of another VoIP provider, the signaling will traverse the VoIP peering exchange but the RTP media stream will most certainly find a more direct path between one another. In the event that the two parties have broadband cable service, then the RTP stream will follow the best IP path between the two broadband cable companies while the signaling will route between the two VoIP providers.

Current VoIP peering options include ENUM and SIP. ENUM provides for a lightweight DNS mapping between an E.164 phone number and an endpoint IP address. ENUM provides for an excellent method for VoIP providers to advertise their ported numbers to other VoIP operators. Some limitations of ENUM include lack of scalability interoperating within a global, multi-homed environment whereby gateways or session-border controllers operate in multiple facilities. Another limitation is that ENUM does not include any signaling or state information such that if a network becomes unavailable or out of capacity, ENUM does not failover. Finally, ENUM is not necessarily implemented in all vendors’ equipment, such as Cisco and SONUS.

SIP, however, provides for two HTTP-like methods: redirect and proxy. A SIP redirect functions much like ENUM yet SIP provides rich information such as codec information to ensure compatible RTP. SIP proxy, on the other hand, provides end-to-end signaling with failover capability, ASR/ACD monitoring and CDR generation. Since SIP peering effectively builds IP trunks between providers, least cost routing on a per peer basis remains an intrinsic function. Third party services such as E911, voicemail and PBX (News - Alert) services integrate easily with SIP within a larger value-add ecosystem.

The early days of the Internet cobbled together various networks using a diverse set of interconnection technologies and methods. Within a few years after commercialization, BGP-4 became the de-facto peering standard whereby large ISPs peered directly between one another in a fully distributed and resilient manner. While the IETF has morphed BGP into TRIPS as an E.164 VoIP peering protocol, this RFC has reached little consensus and seems to be dead.

Finally, many directory-based systems that support alias addressing such as with AIM, Skype, Yahoo and MSN already support SIP based communication between one another. Potentially, integrated clients like Trillian may merge together the large IM networks, as I see little hope in them interoperating on their own. It is for this reason that I foresee enterprise messaging and presence-management systems to become the next big integration component with corporate directories. Doing so allows for a domain-based addressing method so that I can be reached as
eric@heidengroup.com, and whereby a simple query of my DNS server will provide location information for my presence server, its SIP service capabilities, and my availability.

The end game consists of a distributed, broadband-to-broadband capability with enterprise-level control. Just like all Internet communications, voice calls will be free. Features will include rich, policy management whereby I configure who may ring my desk, mobile, and even home phone. Integration between my PC and telephony system will provide for a host of value-added capabilities and control along with integrated video and collaboration tools. Of course, the PSTN will remain (as do postage stamps) but the next generation of computer integrated telephony will quickly change voice communications into an incredibly practical enterprise resource.

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Eric Dean is a principal consultant with the Heiden Group in Reston, Va, He can be reached at (703) 715-4990.

With more than 15 years in the communications industry, Dean has worked with leading Internet-based companies in designing, building and operating carrier-class networks and enterprise architectures. He specializes in developing new products and building high-growth, next-generation technologies and services. Specific to the VoIP industry, Dean has implemented IP-based telephony system within the government space, enterprise markets, and interoperable carrier systems. Always active within the standard's community combined while remaining on the cutting edge of latest vendor advancements, Dean applies practical best-practice principles to business solutions.

Dean’s experience ranges from large-scale IP/MPLS-based global Internet engineering to small-business systems integration. Most recently, he co-developed a SIP-based, carrier-grade VoIP peering platform incorporating ENUM support for colocated VoIP operators within the renown One Wilshire carrier hotel in Los Angeles.

Dean holds a master's degree in telecommunications from the University of Maryland and a degree in electrical engineering from George Mason University. He is a Cisco certified internetworking expert (#2708) and a certified Cisco Systems (News - Alert) instructor for IP- and VoIP-based products.

VoIP


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