January 2004

Session Controllers -- Enabling Service Provider Opportunities In IP Telephony

BY DAN DEARING

Carrier competition is at a premium, with focused attention on the IP telephony market. As carriers seek new sources of revenue, they are forced to adopt more flexible business models that include peering with their carrier partners and enterprise customers. In addition, carriers are seeking more dynamic applications to grow revenues and to act as differentiators. Such applications will need to be scalable and flexible to enable the delivery of VoIP services directly to the enterprise. Within this environment, carriers must meet several technical challenges to fulfill their business objectives. Specifically, while the use of VoIP technologies reduces operating costs and provides new revenue bearing applications, it also creates new issues of security, interworking and multi-vendor interoperability. Fortunately, a new breed of networking platforms called session controllers solves these complexities and enables a diverse set of VoIP services and applications.

A session controller is a new breed of networking technology that provides layer 5 routing and control to manage real time traffic flows between IP networks. Session controllers address issues of network security, signaling interoperability, call admission control, service quality and session routing. Typically these functions are distributed between session controllers deployed at different points within a carrier network. Edge Session Controllers (ESC), deployed at the edge of the network, control a carrier’s connections to their enterprise customers and service provider partners. In this capacity, the ESC provides network security, call admission control (CAC) and signaling interoperability. Core Session Controllers (CSC) are deployed within the core of the network. They are responsible for session routing and a broader level of call admission control (e.g., policies that may limit the total number of calls, which may originate from a carrier partner via multiple ingress points).

Carriers and service providers can use this new networking technology to securely deliver both basic and value-added services via IP in a very simple and cost effective way. Within this IP networking environment, session controllers and softswitch technologies must co-exist to bridge between the PSTN and emerging IP-based networks. Session controllers treat the softswitch and its associated media gateways as an IP endpoint that can service voice calls. The softswitch controls voice-based media gateways and the routing of voice calls between the PSTN and IP network. Session controllers are focused on a much broader variety of real time services (e.g., video, multimedia, IM) that are delivered via packet networks and use the SIP and H.323 signaling protocols. Together they provide the foundation for basic voice services and enable carriers to use IP to expand their service reach and breadth.

Early adopters of session controller technology include voice wholesale carriers, like ITXC and Wavecrest, who have built their networks from the ground up using IP. Traditionally, these types of carriers have used TDM to connect with other partner carriers. This requires a pair of media gateways linked in a back-to-back configuration to convert voice traffic from VoIP to TDM and back to VoIP again. TDM normalizes signaling and media traffic and secures the carrier network since all IP sessions are effectively terminated at the network edge. However, costly DSPs are required on the media gateways to do this conversion. Here lies the opportunity for session controllers since they solve security and interoperability issues and thus enable carriers to use IP to interconnect their networks. Because DSP resources are no longer needed to perform the conversion to TDM, a carrier can reduce the CAPEX cost of their partner interconnects by 50 percent to 80 percent depending on the capacity of the connection.

Session controllers also provide carriers with much needed OPEX savings since their use simplifies peering and reduces turn-up time and cost -- one large VoIP carrier estimates that VoIP peering is one sixth the cost of traditional TDM approaches. In addition to savings provided by the use of session controllers deployed at the edge of the network, the CSC also provides savings in the core of the network. The stateful device provides centralized collection of call detail records and enables carriers to recover revenues associated with lost call detail records that result when CDRs are collected by edge devices. The CSC’s sophisticated routing engine also maximizes route revenue and enforces agreed upon business terms to deliver the highest possible revenues and margins for carrier to carrier services.

Session controllers also create opportunity for carriers delivering services to the enterprise. In recent research with global service providers on their plans for packet voice, the Yankee Group found that 73 percent of wireline respondents indicated a need to peer IP PBXs into their networks. Using session controllers, carriers can fulfill this need and offer their enterprise customers a broader menu of retail voice services including voice VPNs, local and long-distance termination, and hosted PBX (or IP Centrex). For basic voice services, session controllers enable carriers to control, route, and manage enterprise VoIP traffic in a secure and seamless manner. Because IP is the delivery mechanism, the carrier can offer the enterprise converged services for both their voice and data needs resulting in lower costs for the customer. This convergence allows the enterprise to use IP trunks instead of traditional TDM trunks to connect with the PSTN and provides a greater ROI for their IP PBX.

Session controllers give carriers the flexibility to offer managed connectivity for enterprises using IP PBXs and those that prefer a hosted PBX solution. The proliferation of VoIP endpoints (e.g., SIP phones, H.323 PBX access gateways, etc.) within the enterprise has forced the carrier to confront VoIP traffic on the access network. H.323-based IP PBX solutions have been widely deployed by the enterprise while service providers are deploying application infrastructure to prepare themselves for the flexible and collaborative applications being developed for SIP. This is typified by the deployment of hosted PBX applications for the small and medium sized enterprise market using SIP-based application servers. Carriers must adapt to this environment using a technology strategy for H.323 to SIP interworking which ensures maximum service reach to any endpoint.

The SIP/H.323 interworking function of the ESC enables the carrier to interface with a broader set of IP PBX implementations and abstracts call signaling from other policies, such as CAC and call routing. This capability is a key service enabler that integrates traditional voice services via H.323- or SIP-based media gateways, hosted SIP-based voice services, and multiprotocol access services for IP PBXs and IP enabled PBXs. Carriers can use the CAC capability of the ESC and CSC to deliver access services to their enterprise customers based on virtual trunks. Because the ESC is able to control the number of ingress and egress VoIP calls to the enterprise, carriers have the flexibility to offer their customers two-way, inbound and outbound trunks based on the enterprise’s call flow. The number of virtual trunk lines and their direction can easily be changed by the carrier’s operational staff. Operationally, the ESC provides carriers with a flexible mechanism for access control and real time private-to-public IP network address translation (NAT). These capabilities support NAT traversal, topology hiding, route enforcement, and the regulation of bandwidth consumption to manage multimedia flows across the network boundary between the carrier and enterprise.

While IP PBXs are ideal for larger corporations, with IT staff accustomed to operate and maintain them with other IT solutions, IP Centrex or hosted PBX services offer SME and SOHO users the benefits of IP telephony without the operational overhead. Many IP Centrex subscribers use broadband connectivity to their locations. The challenge within this environment is NAT traversal or the ability to provide secure connectivity to subscribers behind NAT devices and firewalls. NAT devices prevent two-way voice and other methods of real-time communications, because they lack the intelligence to make the necessary address translations deep inside VoIP signaling packets. Session controllers solve this problem using a variety of client/server techniques such as STUN (Simple Traversal of UDP through Network Address Translators) for SIP applications and other proprietary tunneling techniques that address a broader set of signaling protocols.

While the ESC provides interoperability between IP PBXs and hosted PBXs, the CSC provides a flexible and sophisticated route engine that enables the interconnection of legacy PBXs (via media gateways), IP PBXs, and hosted PBXs into a single cohesive service. This programmable route engine enables flexible end user calling plans through the use of call treatment facilities and call routing by time-of-day, device capacity, device utilization, and device priority. The CSC also provides service- level CAC polices which are enforced across multiple devices (e.g., gateways and ESCs) on a customer basis. With these capabilities, carriers enable their enterprise customers to leverage both IP PBXs and hosted PBX solutions to meet the unique requirements of their office locations.

Session controllers are needed to overcome technical barriers in order for carriers to directly interconnect with the disparate IP networks used by their enterprise customers and carrier partners. Using this new networking technology, carriers can reduce the CAPEX and OPEX costs needed to interconnect with other VoIP networks while also expanding their current service reach and revenues. More importantly, carriers can establish a robust foundation for other real time services and value-added applications, such as video and teleconferencing, for greater revenues and higher margins.

Dan Dearing is vice president of marketing at NextOne Communications. NexTone is a leading provider of session controllers for the delivery of real-time services over packet networks, such as voice over IP. NexTone solutions enable carriers and service providers to securely, simply and cost-effectively interconnect networks.

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