Session Initiation Protocol (SIP) goes beyond other VoIP protocols through its power to connect multi-media devices with each other and with existing Web assets. While it will lower telecom operating costs, ultimately, SIP is the key to bringing powerful telephony functions – unified communications, click-to-communicate, intelligent find-me, multi-media collaboration – to desktop, mobile endpoints, as well as business applications.
The good news is that it’s possible to gain immediate benefits from SIP without a complete overhaul of your existing Avaya telecom infrastructure. SIP migration can be accomplished through a series of steps that deliver advantages at each point.
Avaya’s three steps is a low-risk approach that doesn’t involve a significant initial investment. When you are ready, you can extend SIP beyond the edge, to the telecom core and eventually to user desktop and device endpoints.
Step 1: Single SIP Trunk for VoIP and Data Services
Overwhelmed by SIP? There’s a low-overhead, low-cost way to take advantage of Session Initiation Protocol that doesn’t involve significant changes to your current telecom environment. The answer comes from the external VoIP service providers that are SIP compliant. Taking advantage of this step is simple: your business connects the Avaya Communication Manager trunk-side to a service provider’s network interface.
A SIP trunk from a compliant service provider delivers important cost benefits to your business. With SIP trunking, inter-office calls are routed by the service provider over its network. As long as the call avoids the local carrier’s network and remains on the provider’s – “on net” – no per-minute toll charges are involved, resulting in lower recurring telecom costs.
For calls that are “off-net” – a call to the public network or PSTN – the SIP service provider, through its VoIP gateways, terminates the outgoing call, translating between SIP and TDM.
The benefits of a SIP trunk extend beyond low-cost voice calls:
- Enhanced services Direct inward dial (DID), toll free numbers, calling cards, and local presence numbers can be quickly provisioned by a SIP service providers and at lower costs than a legacy TDM carrier.
- Date networking SIP service provider generally offer a bundled service arrangement involving data and voice networking between locations and access to the public Internet – usually at a fixed cost per site.
- Equipment Consolidation because data and voice are carried on the same circuit to the service provider, enterprises have the opportunity to consolidate their networking equipment for a reduced infrastructure maintenance and investment.
- Scalability SIP trunking requires only one SIP connection per location to the provider’s network. This contrasts with traditional TDM voice networking where separate point-to-point circuits are associated with each remote office connection. Growing businesses can achieve significant savings because per location network connection costs remain fixed —the cost of the single SIP connection covers complete network access.
Looking ahead, SIP peering will allow service providers to offer on-net calling to other businesses on the provider’s SIP network. Result: calls that were previously treated as long distance avoid associated toll charges. And advanced features like presence will be available between businesses – for example, buyers are notified of the presence status of key suppliers.
Step 2: SIP Servers and Applications at the Platform Core
While involving the installation of customer premises software, this phase has the advantage of limiting SIP infrastructure to the enterprise’s core. Existing Avaya IP endpoints (the Avaya 4600 series) and Avaya’s call processing hardware remain in place. SIP comes into play in the form of SIP-centric servers that take on application roles.
Avaya has SIP-enabled many longer-standing products, such as Avaya Modular Messaging, Avaya Voice Portal, and Avaya Application Enablement Services. Besides the desirable goal of supporting end-to-end SIP transit, SIP enabling a server has other benefits. In the case of Avaya Modular Messaging, SIP improves server performance and scalability through a load balancing scheme.
Focus on Avaya Voice Portal
The Avaya Voice Portal 4.0 is a VoiceXML- based server on which can be built basic IVR functions as well as more advanced speech recognition applications. Using a standardized XML subset – Voice Portal is compliant with VoiceXML 2.0 – developers craft scripts to implement applications that directly convert existing web HTML content to speech.
Voice Portal is a good example of a SIP-enabled server that delivers immediate benefits to businesses without a “fork-lift” of existing equipment and software.
When coupled with SIP trunking, Avaya Voice Portal can directly handle calls from the SIP service provider. Voice Portal collects DTMF and speech input from customers and then queries Web and IT assets to generate dynamic speech prompts.
Because calls are terminated on the edge of the network, there’s decreased use of internal call centre network and hardware assets. For customer interactions that do require an agent, Voice Portal can work in conjunction with Avaya Interaction Centre (without changing ACD vectors) to route calls to appropriate knowledge resources anywhere on the enterprise network.
Step 3: SIP Endpoints
The final phase of the migration involves the endpoint device. Within Avaya’s endpoint framework, SIP upgrades for many existing IP phone can be made through a downloadable firmware upgrade. And because Avaya Communication Manager supports interworking between H.323 and SIP, existing Avaya phone users can communicate with newly enabled SIP phones in true peer-to-peer fashion.
The case for moving to SIP at the endpoint level extends beyond inexpensive SIP devices or softphones. SIP brings to the telecom table support for presence, device modalities, and multi-media applications.
Presence and Modality
With the investment in SIP services at the core (step two), SIP presence information “the new dialtone”- can be applied to making more intelligent connection decisions. SIP user preferences, as maintained in the proxy server, specify how calls will be directed and controlled.
Example: a preference can indicate that when a user is working on his digital PDA, he only communicates via text messages. A voice call to the PDA would then automatically initiate an IM session on both caller and callee devices.
Multi-Media
With high-res LCD screens (available on some Avaya one-X deskphones) application can be developed (step two) that combine both voice and Web data.
Example: a technician calls into a self-service application that’s implemented on the Voice Portal. At some point in the interaction, the technician realizes she needs to order a new part. Instead of having to abandon the call, the technician can request that the Voice Portal download the Web form directly to the SIP phone. The Avaya one-X deskphone’s browser pulls the appropriate page from the Web server, allowing the technician to order the part, and then continue the self-service call.

