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Feature Article
August 2004

Session Controllers and the Complexities of Connecting VoIP



Voice over IP (VoIP) is rapidly gaining traction among competitive and incumbent carriers because of the compelling economics of packet-based over circuit-based transmission. While up to 90 percent of international voice traffic today is packetized for at least part of the transmission, more carriers are offering packet voice for local and in-country calling � and consumers are signing up for service. However, for VoIP to meet the full potential of consumer demand, carriers must solve critical interoperability issues that exist because of the way VoIP implementations have evolved in the industry.


To understand these interoperability challenges, one must review the ever-changing protocol standards and their evolution to support more robust applications while also providing operational efficiencies. Session Initiation Protocol (SIP), the Internet Engineering Task Force (IETF) approach for VoIP, is centered on making voice calling more Internet-like. The SIP protocol is modeled after other Internet protocols, such as HTTP and SMTP, where flexibility and simplicity are key attributes. The International Telecommunications Union-Telecommunication Standard-ization Sector (ITU-T), on the other hand, developed H.323 to ensure interoperability among systems that deliver real-time communications over packet-based networks. Despite the fact that H.323 and SIP address similar requirements, the mechanics of how they perform call setup, media negotiation, and call tear down, make them incompatible, preventing direct connectivity between SIP and H.323 endpoints.

Session controllers provide the intelligence to securely interconnect carrier networks with other SIP and H.323-based networks. This capability, SIP/H.323 interworking, is one of the important ways session controllers are helping carriers simplify their networks and profitably introduce and expand VoIP offerings and other real-time services over cost-effective IP infrastructures.

H.323 Versus SIP
The H.323 and SIP protocols were developed with similar intentions, but with different orientations and techniques � to enable real-time services across packet networks.

H.323 came out of the voice-oriented, circuit-switched world. This protocol was designed to support video conferencing among multiple IP networks; it is the first version of the ITU specification, and dates back to the early 1990s. Because H.323 provided some key call-control and gateway-administration functionality, carriers chose the protocol as IP telephony gained market traction. Though interoperability issues created bad buzz for H.323 even early on, there exists a significant base of users. Not being able to afford being left behind as more enterprises moved to VoIP and other IP-based services, incumbent carriers invested heavily in H.323.

Application developers and, especially softswitch vendors, derided H.323 as overly cumbersome and complex, lumbering and memory-hogging � a desperate creation of the Time Division Multiplexing (TDM) world that was ill-suited for the emerging, all-IP future, especially in the case of VoIP. SIP arrived late in the decade, spawned by the IETF and hailed as an enabler of not just of IP telephony, but as a key underpinning of the entire converged, application-oriented, next-generation Internet. The protocol defines methods for initiating, modifying, and terminating interactive user sessions.

Uncovering the need for a new tool
The H.323/SIP eternal battle proved bad for business. Established carriers that heavily invested in H.323-based platforms found it operationally difficult and expensive to interconnect with new providers and enterprises committed to the new SIP standard. Because various SIP-based systems were engineered at different stages in the protocol�s definition, not all SIP-based systems shared a common interpretation of the standard�s details (in the areas of message format, timing, and sequence, for example). The same can be said for H.323-based platforms. The ramifications have caused problems with signaling interoperability, network address translation (NAT), and firewall traversal.

TDM peering, with softswitches and media gateways, gave carriers an approach to overcome the SIP/H.323 interoperability issues with media gateways linked back-to-back, converting voice traffic from VoIP to TDM and then back to VoIP at each carrier interconnect. This solution was a workable but cost-prohibitive strategy, requiring significant operational and capital expenditures. So, in the absence of real SIP/H.323 interworking, the introduction and scaling of VoIP and other real-time service offerings remained largely on hold.

Then came the session controller�

Normalizing at the edge, standardizing in the core
A Layer 5 technology, session controllers enable carriers to move to VoIP peering � far simpler to engineer than Time Division Multiplexing (TDM) peering and reliant on none of the expensive DSP resources. Small carriers offering wholesale VoIP services were the early adopters of session controllers.
Today, as session controllers have evolved, the entire carrier community is making a fundamental shift in the infrastructure model for supporting VoIP and other real-time, packet-based services. The industry�s best-of-breed edge session controllers operate on three planes: signaling, media processing (transcoding among various codecs), and media routing (NAT and firewalling).

Session controllers operating in the network core provide end-to-end signaling capability, acting as call control engines to provide intelligent routing, and traffic management. Edge session controllers operate in points of presence (PoPs) to handle interconnections with enterprise customers� IP Public Branch Exchanges (IP PBXs) or carrier partners� IP networks. The media processing and media routing functions of edge session controllers are as important as signaling compatibility to carriers, because G.711, G.723, and G.729 codecs approach media compression in different ways. The edge session controller negotiates between incompatible codecs.

The softswitch, meanwhile, provides the mechanism for IP-enabling traffic orienting from the circuit-TDM-based, Public Switched Telephone Network (PSTN). Core session controllers treat the softswitch and its associated media gateway as just another IP endpoint. In turn, ports on the expensive TDM backbone switch and media gateways that were previously used for VoIP-to-VoIP traffic can be harvested and cost-effectively re-assigned for TDM-to-VoIP traffic.

Making the transition from TDM peering with softswitches to VoIP peering with session controllers alone can cut a carrier�s costs by 50 to 80 percent (depending on the capacity of the network interconnection), spurring broad-scale rollout of more competitively priced services. With session controllers delivering true SIP/H.323 interworking and media processing, carriers are freed to take advantage of the capital and operational cost efficiencies of standardizing their VoIP core technologies. Regardless of which flavor of H.323 or SIP or media compression used in the carrier�s own network core, the market�s leading edge session controllers enable cost-effective interconnection with VocalTec, Clarent, Cisco or any other standards-based IP network.

Moving to end-to-end real-time control

With session controllers shouldering the responsibilities for signaling interoperability, network security, call admission control, and service quality control at the network edge, carriers are now increasingly realizing their benefits in core deployments.

The technology�s very granular and highly programmable routing capabilities are especially valuable in the network operations center (NOC). Session controllers can assess the multitude of peering points a carrier might have with carrier partners and enterprise customers and leverage policies in routing and managing authorized traffic (e.g., there might be a policy to limit the amount of calls originating from a particular carrier partner via several ingress points.) The session controller ensures that traffic is routed to optimum peering points, boosting call-completion rates.
VoIP billing is another responsibility that is more frequently relying on core session controllers. Demand for VoIP and other real-time, packet-based services (video hosting and other multimedia opportunities that will ultimately prove country-specific) is intensifying as prices drop and availability widens. Continuing to seek new levels of network simplicity, carriers are coming to regard session control as a multidimensional technology delivering cost-effective control end-to-end across their IP networks. As a result, core session control is rapidly increasing in popularity.

The promising equation of Internet telephony � basing voice services on less-expensive equipment that would equal lower prices and greater demand � failed to add up to wide-scale rollout until recently. The incompatibilities among SIP- and H.323-based VoIP systems comprised one of the primary obstacles, but those complexities have been solved with session controllers normalizing traffic at the network edge. Via VoIP peering, interconnection is seamless, so carriers can cost-effectively expand infrastructures and service footprints.

For the foreseeable future, there must be accommodation for both SIP and H.323 in carrier IP networks. Session controllers protect space for both of the protocols, through SIP/H.323 interworking.


Daniel C. Dearing is vice president, marketing for NexTone Communications. For more information, please visit the company online at www.nextone.com

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