Solving NGN challenges with SIP signaling

Vince Lesch, Tekelec
15 Sep 2009

Since its debut in 1995, VoIP has evolved from a hobbyist’s pursuit to a mainstream network technology. Operators around the globe are deploying next-generation VoIP networks. Providers see VoIP and SIP-based networks as an opportunity to cut their operating expenses and boost their bottom line with new revenue.

However, as VoIP and SIP traffic and applications grow, so do the requirements on the underlying network. Operators are discovering that their next-generation networks (NGNs), built on a voice-centric, softswitch architecture, don’t have the scalability and flexibility to support multimedia services, access independence or network growth.

The missing layer
In the rush to get VoIP up and running, many providers took a short cut. They deployed their NGNs as a loose collection of elements interconnected by direct signaling links. Unlike signaling system 7 (SS7) and Internet protocol multimedia subsystem (IMS) networks, there’s no signaling and session layer to handle application-layer tasks.

From a signaling perspective, each NGN element must handle all application-layer related tasks like routing, traffic management, redundancy and service implementation. Every possible route must be defined at each network element, creating a spider web of logically connected SIP nodes.

A better approach is to deploy a SIP signaling router (SSR) – a SIP proxy with enhanced routing capabilities -- to centralize layer-5 SIP routing into the network core. The SSR creates a signaling and session framework that relieves the endpoints of session management duties. The resulting architecture allows the NGN to grow systematically and creates a flexible framework that enables:

  • Enhanced application server (AS) selection: The tight coupling between SIP endpoints like SIP phones and SIP application servers creates a challenge for many operators. Any changes made to the physical network, such as adding a new application server, have a direct impact on the way SIP phones access service. The SSR shields the endpoints from changes in the physical network through a process called “abstraction” in which the phones are decoupled from direct knowledge of the complex and changing network; SIP phones just have to be configured with a single abstract address. Endpoints send requests to the SSR, which resolves the address to the appropriate SIP AS platform and routes the request to that platform.
  • SIP trunking: Softswitch-based, SIP trunking solutions, which are built on a “per-connection” cost model, can become costly very quickly. And, since softswitches usually are deployed with the switch vendor’s choice of application server, it’s difficult to gain the economy of a “best-of-breed” solution. By implementing an SSR, operators can use a session-based approach to provide fixed-line services to enterprise customers. The SSR routes on-net calls between IP PBXs and off-net calls through a PSTN gateway to local and long-distance fixed numbers. The resulting architecture creates a volume-based cost structure and reduces costs by allowing operators to select “best-of-breed” application servers.


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