Enhanced service delivery in the packet voice network

May 1, 2002 12:00 PM, Grant Henderson and Kenneth Osowski

An industrywide movement to deploy IP-based, converged network infrastructure is now opening up new markets and revenue opportunities for traditional service providers with their existing legacy networks and for emerging carriers with their IP-based, next-generation networks (Figure 1). The first move made by traditional carriers to deploy converged network elements has been for them to use public network/IP media gateways to perform Internet offload functions. Internet offload has enabled these carriers to defer expensive upgrades on their TDM switches by diverting long hold time, modem-based Internet calls onto data networks. Emerging carriers along with some traditional carriers (both wireline and wireless) are using media gateways to convert public network voice calls to "packetized" voice and route these calls to lower-cost IP network infrastructure (IP trunking).

Both of these converged network deployment scenarios are enabling carriers to significantly reduce costs in their networks. However, service providers clearly want to enable service revenues with this infrastructure, to make the business case for moving to next-generation infrastructure much more compelling. In fact, current buying cycles for media gateways often include the carriers' desire to deploy proven revenue-generating services such as calling card services, conference calling, Centrex and voice mail services from the outset. These solutions are incorporating an IP-based service architecture that includes IP media servers and session initiation protocol (SIP) based application servers connected to media gateways. This allows service providers to deploy proven, core network services today that don't require expensive TDM-based service platforms. And as packet telephony technology reaches out to the access part of the network, this service architecture becomes the basis for developing innovative new services that only an IP-based services model can enable.

The move to a pure-IP services infrastructure can help service providers reach their goal to increase revenues and lower infrastructure costs, bringing them a higher level of profitability. They will then be making future-proof technology choices that position their voice networks into a packet-centric future.

The traditional approach used to deliver high-value voice services to public networks has been through the deployment of service nodes. These platforms included dedicated, purpose-built hardware and software capable of deploying services such as voice mail, find me/follow me, unified messaging, prepaid calling card, interactive voice response, 800 number services, and conference calling. Most of these service node product offerings were designed to deliver only one of these services at a time using a proprietary design that made it difficult to deploy multiple services. If service providers had a requirement for new services, service node vendors typically developed customer-specific software releases that could be very costly. This software development process typically took six to 18 months since service logic was tightly integrated with internal signaling subsystems, proprietary DSP hardware, and capacities of underlying hardware components.

A new approach has emerged to deploy next-generation service architectures using IP-based technologies. This approach decomposes the traditional TDM service node into hardware and software components separated into distinct functional subsystems and interfaces based on open communications protocols--SIP, media gateway control protocol (MGCP) and real-time transport protocol (RTP)--and commercially available computer systems, databases, and Web servers. Service logic is now separated from the hardware and runs on what is generically called an Application Server. Newly developed service creation environments can now create service logic that is separated from underlying hardware platform dependencies, providing a clear delineation of hardware and software components allowing service providers to specify best-of-breed components. Cost savings and capacity benefits derived from using commercially available components can be passed directly on to service providers.

Another major component of this next-generation services architecture is the IP-based media server. This component takes DSP logic running deep inside traditional service nodes and makes these functions available to the service logic execution environment running on the application server. These functions include interactive voice response capabilities such as playing prompts and collecting DTMF digits, automatic speech recognition and bridging calls together for conferencing applications, as well as recording greetings, announcements and messages to a server.

As call flow logic executes, the application server requests these functions when needed from the media server using either SIP or MGCP protocols. Not only does this decomposed service node model speed up application development, it allows these components to scale independently so that new services and capacity can be easily added.

So how do these components fit into existing or greenfield networks? The simple answer is to add a media gateway or voice router that supports SIP signaling and connect the application servers and media servers behind it.

For emerging carriers implementing greenfield networks that interface to the public network, media gateways are often deployed to perform IP trunking that provides connectivity to the public network and allows for transport of voice traffic across IP networks (Figure 2). This creates a cost-effective approach for them to build out brand new networks using IP network infrastructure. But it also enables them to incrementally add IP voice services to this network by "plugging" in SIP-based application servers and IP media servers to the converged network infrastructure. Not only does this save service providers significant costs vs. implementing TDM service nodes, it also gives them speed-to-market and services definition flexibility. This is especially important to emerging carriers who are competing with incumbent carriers on the basis of lower pricing and service offering differentiation.

For traditional carriers with a large investment in TDM networks and service nodes, IP voice services can be deployed today to replace TDM service nodes, creating a bridge between the public network and an IP network that enables these systems to be accessed by service subscribers on any wireline or wireless phone. Because this offloads voice traffic specific to services on to an IP-based telephony network, this deployment scenario is called "enhanced services offload" (Figure 3). This approach allows service providers to either replace legacy TDM-based services or cap-and-grow onto a cost-effective services architecture.

In order to see how these components work together, we can run through simple call flow scenarios for both prepaid calling card and conference calling. In the case of prepaid calling card, a subscriber would typically call an 800 number provisioned on the media gateway to access the service. Based upon the number called, the media gateway "forwards" the call to the application server by issuing a SIP invite request. The application server then starts prepaid service logic that plays an introductory service greeting and prompts the subscriber to enter their PIN code. During this part of the call flow, the application server asks the media server (usually under MGCP control) to play the prompts ("Please enter your PIN code") and collects the subscriber's PIN code that was entered using DTMF input on the subscriber's phone. The "bearer channel" or voice path that enables the subscriber to hear the voice prompts is established between the media gateway and media server as RTP streams.

For conference calling, subscribers call the 800 number on the media gateway to access the conferencing service. Again, subscribers are authenticated as in the prepaid calling card service example. The application server creates an active call session for each participant in the conference call and maintains the state of the conference. All of the audio from these call sessions is bridged on the media server, and RTP streams are maintained for each participant for the duration of the call. The application server can be processing both of these services at the same time, allowing the service provider to consolidate resources and manage multiple services in a single service environment.

So what benefits can service providers achieve today by deploying this IP-based services architecture in their networks? 

• Eliminate dedicated TDM trunking facilities and overlay switching fabric

• Eliminate hairpinning of calls saving on expensive TDM VRU ports for prepaid calling, find me/follow me, 800 number services and other applications. Hairpinning occurs when a subscriber calls into a service access number and places an outbound call through the service. Even though this would happen on the media gateway, it would not happen on the IP voice services platform

• Run multiple services in a unified IP-based services architecture

• Deliver services to both consumers and businesses on the same service platforms

• Install brand new high-growth services such as conference calling that were previously too resource and capex-intensive in TDM environments

• Achieve more cost-effective market trials allowing for regionwide or new geographic market deployments to more broadly test new services

• Deploy cost-effective and scalable media resources that can be used across multiple services

• Start to deploy IP-based, next generation, packet telephony protocols

• See return on investment (ROI) that is less than 12 months even for deployment of a single service

• Use SCEs that are tied to open protocols and are not vendor-specific.

Conference calling as an example has traditionally been expensive to deploy using TDM approaches. Typical TDM conference bridges today might consume an entire 19-inch rack with maybe a 1000 ports. IP media servers today can physically scale up to 18,000 ports in a single shelf. This greatly reduces carrier costs for network connectivity, maintenance and manpower. Given the IP nature of these new services platforms, Internet integration for subscriber control of the conferencing service is much simpler to integrate architecturally.

Implementing IP-based voice services today makes a great deal of sense, even if the carrier has not yet made a networkwide decision to implement SIP-enabled softswitches and gateways. The business case for doing a service at a time can allow carriers to go after new market opportunities and achieve ROIs that are less than a year.

Grant Henderson is Executive VP for Convedia Corporation. Kenneth Osowski is VP of Product Management for Pactolus Communication Software.

Visit Convedia Corp. and Pactolus Communication Software online.