SIP Signaling

Interesting article from http://www.totaltele.com/view.aspx?C=0&ID=443303

Creating Flexible Next-generation Networks with SIP Signaling

Monday 16 February 2009

Creating Flexible Next-generation Networks with SIP SignalingNext-generation network (NGN) deployments are on the rise. Mobile and fixed line operators are moving to Internet protocol (IP) technology to reduce their transport costs, deliver voice-over-IP (VoIP) services to consumers and enterprise customers, and improve the delivery and management of new multimedia services.

As IP and session initiation protocol (SIP)-based services proliferate, so do the requirements placed on the network to support them. Operators are discovering a downside to their softswitch-based NGNs; they lack the scalability and flexibility to support multimedia services, access independence or network growth.

The Missing LayerIn the push to get their NGN networks up and running, many providers took a short cut. They deployed their VoIP networks as a loose collection of elements interconnected by direct signaling links. Unlike SS7 and Internet protocol multimedia subsystem (IMS), which are hierarchical networks, there’s no signaling and session layer to handle application-layer tasks. From a signaling perspective, each network element must handle all application-layer related tasks. Every possible route must be defined at each network element, creating a spider web of logically connected SIP nodes.

Creating a Session-Control Layer in NGNsCreating a signaling and session framework in the core NGN network avoids the pitfalls created by point-to-point routing. By deploying a SIP signaling router (SSR) - a SIP proxy with enhanced routing capabilities - operators can centralize layer-5 SIP routing in the core network and relieve endpoints of session-management tasks. The resulting architecture creates a flexible framework that enables the following use cases:
Enhanced application server (AS) selection: The tight coupling between endpoints like SIP phones and SIP application servers creates a challenge for many operators. 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 by decoupling them from direct knowledge of the changing network. The 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 a 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 public switched telephone network (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.
SIP number portability (NP): For operators with a SIP-trunking infrastructure, performing number portability for VoIP calls can present a particular problem. They can simply “dump” the calls on the PSTN gateway if there’s enough intelligent network capacity and the terminating network is time division multiplexing (TDM). But, if the terminating number is another IP PBX or belongs to a VoIP operator, the call must be shuttled from VoIP to TDM and back to VoIP again. Running pure VoIP calls over TDM wastes gateway capacity and degrades voice quality. Another alternative is to replicate a number portability solution in the SIP domain, but that’s a costly approach. Using the SS7 access feature of the SSR, operators can make TDM-NP available to the SIP network. This capability allows the SSR to augment its routing capabilities with SS7 data. Pure VoIP calls don’t have to be shuttled over the TDM network to perform NP, which maintains voice quality and saves PSTN gateway capacity.

Centralized SIP proxy: Expanding NGN networks requires the addition of new softswitches. Each new piece of equipment must be provisioned with the routing entries for all of the existing softswitches, and existing softswitches must be updated with the routing entries for new equipment. Route management, which is based on pre-defined SIP trunks, becomes increasingly complex as the network expands. Service and subscriber data are tightly coupled with the softswitch, making it difficult to change an existing service or add new applications uniformly. The SSR deployed as a SIP proxy creates a SIP-based reference architecture over the existing network. Calls are routed by default from the softswitch to the SSR. The SSR makes layer-5 SIP routing decisions based on advanced routing algorithms and forwards the request to the appropriate SIP destination.

Specialized SIP proxy: As operators consolidate their networks, many are discovering that softswitches supplied by different vendors are unable to establish sessions. That’s because each vendor uses a different SIP implementation. As long as an operator deploys equipment from a single vendor, there’s no problem. But, when equipment from another vendor is introduced, interoperability problems arise. The issue can be resolved with a customized solution, but that’s an expensive route to take. The SSR creates an architectural solution that is independent of the endpoints and eliminates interoperability problems. Deployed in the signaling layer, the SSR serves as a SIP proxy. It routes SIP traffic between the softswitches and serves as a mediation point between them, “fixing” protocol variations on the fly.

SummaryHaving softswitches and other endpoints perform layer-5 session management may be sufficient for fairly small deployments and simple management tasks. But, as the network expands, the lack of a capable session framework introduces a host of network issues. A suitable session framework offloads signaling and session tasks from the edge next-gen elements to enable efficient network expansion. Just as core routers are used to minimize the routing burden on IP endpoints, layer-5 SIP routing reduces the burden on endpoints by centralizing session management tasks at the network core. The resulting architecture can expand systematically to support VoIP subscriber growth, deliver advanced multimedia services and create the foundation for future technologies and services.

About TekelecFound at the heart of most global networks, Tekelec’s market-leading, mission-critical, high-performance network solutions enable the secure and instant delivery of calls and text messages for more than one billion mobile and fixed-line subscribers. The company’s session management solutions allow telecom operators to manage diverse applications, devices, technologies and protocols, across existing and evolving networks, to meet the demands of today’s consumer. Tekelec uniquely ensures telecom operators have a clear migration path to SIP-based IP networks, and whatever comes next, with the flexibility to deploy solutions at a pace dictated by their business needs. For more information, please visit www.tekelec.com.