New Media

Offering telephony service is a journey, not a destination. Along the way, that journey will take cable operators from circuit-switched to Internet protocol-based networks. No one disputes that the future architecture of cable's competitive networks will be packet-based, but the shift to packet-based telephony over cable will not occur overnight-getting there will take more time than most service providers realize. Three key facts support this point: the feature and functionality demands of consumers, the need for consistently high-quality voice connections and the history of technology introductions in the telecommunications industry.

In cable networks, both circuit-switched and IP technologies are overlaid onto a hybrid fiber/coax (HFC) network, which uses optical fiber for signal distribution along trunks or between cable headends. Fiber's resistance to noise and low signal attenuation drove its acceptance as a preferred trunk medium in cable networks. As fiber became more prevalent, its costs decreased, which helped extend it further into the cable network.

Today, a common configuration for larger service providers is a star-star-tree topology. In this design, fiber extends from the headend to hubs in the first star layer, and then from hubs to fiber nodes in a second star layer. The final layer begins with optical to electrical conversion at the fiber nodes and continues with sets of coax tree branches from the fiber node to the subscriber.

Adding circuit-switched telephony to an HFC network requires three network elements: a telephony switch, a network interface unit (NIU) and a host digital terminal (HDT). Their placement in an HFC cable network is shown in Figure 1.

The functions of a telephony switch can be grouped into three categories: call processing, call routing and feature provisioning. Call processing and call routing set up a path through the public network between parties when a call is initiated. Enhanced capabilities such as call screening are provided from the digital switch through the HFC telephony network. Associating these feature capabilities with an originating line is an important part of the switch's function. Switch software can also provide call forwarding, call transfer, call waiting and call conferencing.

As part of the call processing function, the telephony switch can communicate with the SS7 network and its databases. This capability is a powerful enhancement to the switch software and provides for calling number identification and number portability.

In telephony, the NIU is the demarcation point between subscriber-owned equipment and equipment owned and maintained by the service provider. Single-family NIUs typically provide one CATV line and two telephony lines, while multidwelling NIUs can provide for 12 or more subscriber terminations. Both types of NIUs may also include an Ethernet port for delivering high-speed data capabilities to the subscriber.

The NIU has outgrown its traditional role as a passive termination point. Although it still terminates the cable company's coax plant at the customer premises, it must also provide:

* Twisted pair termination for telephony.

* Analog-to-digital conversion and vice versa for voice telephony.

* Packetization of digital information.

* RF modem.

* Diagnostics.

* Dial tone and ring generation.

As the interface between a cable distribution system and the telephony switch, the HDT acts as a digital multiplexer. It provides T-1/E-1 links to the telephony switch at 1.544 Mb/s and accepts 64 kb/s digital signals from lines on the subscriber side, usually in a T-1/E-1 format.

Most vendors have designed the HDTs with an open interface to the telephony switch, allowing the service provider to choose different vendors for the switch and the HDT. On the subscriber side, however, the connection to the NIU is proprietary, requiring the cable operator to purchase both the HDTs and NIUs from the same vendor. Having an open interface to the switch also enables a cable operator to obtain telephony switching from another company through alliances or leasing agreements, so the operator doesn't need its own digital switch in the early stages of telephony offerings.

IP telephony architecture When comparing IP with circuit-switched telephony, the key words to remember are transport and routing. Unlike circuit-switched telephony, IP telephony architecture does not have a standard mechanism to provide the per-line call enhancement features found in a telephony switch.

The dominant standard for IP telephony, ITU H.323, specifies four functional components, or building blocks, for a network that carries IP telephony traffic: terminals, gateways, gatekeepers and multipoint control units (MCUs). These components may be separate pieces of hardware and software, or they may be incorporated within an existing network element. Figure 2 shows the placement of these components in a typical IP telephony architecture.

Terminals are the endpoints of the network. Although most terminals are associated with hardware such as a PC or a telephone set, IP telephony terminals are defined by their software. H.323 specifies the modes of operation for different audio, video and data terminals to work together. Through various other elements of the H.323 standard, these terminals must negotiate channel usage and capabilities, provide signaling and call setup, communicate with a gatekeeper and allow for sequencing audio and video packets.

Gateways are the keys to IP telephony, the bridge between the public network and the IP network. Architecturally, a gateway can be connected to either the access line side or the trunk side of a switch. Gateways perform six basic functions:

* Address translations. The gateway performs translations between an IP address for routing through the Internet and a dialed telephone number.

* Call connection. This includes all call setup signaling and negotiation.

* Analog-to-digital conversion.

* Demodulation. This function is associated with determining whether a digital call is a voice or fax call, and demodulating a fax signal back to the original digital fax format before packetization.

* Compression. This occurs from 64 kb/s to a lower bandwidth, typically 8 kb/s.

* Decompression/remodulation. This is the conversion of incoming packets to the appropriate analog or digital telephony interface such as POTS, ISDN or T-1.

Gatekeepers are optional under H.323. When present, they assist the gateway in processing and routing calls. Gatekeepers provide:

* Address translation. This is the same function as described for gateways. When a gatekeeper is part of the network, the function resides in the gatekeeper.

* Admissions control. The gatekeeper processes messages that indicate whether a caller is authorized to use the network.

* Bandwidth control. The gatekeeper must process messages from terminals requesting specific amounts of bandwidth.

* Zone management. Terminals are assigned to gatekeepers within specific zones. The gatekeeper must provide the above three functions to all terminals in its zone.

In addition, a gatekeeper may provide optional signaling, call authorization, call management and bandwidth management functions.

Multipoint control units (MCUs) enable conference calls between three or more endpoint terminals. Under H.323, an MCU consists of a required multipoint controller and other optional multipoint processors.

Technical considerations Two major issues must be considered when a cable operator evaluates IP as the technology of choice for providing telephony service.

First, IP telephony is a transport technology, so it is currently better suited for network rather than access applications. Presently, IP does not have a standard way to provide feature-rich calling capabilities to subscribers. Call detail recording and billing are also in the evolutionary stages, as is communication between gatekeepers and the SS7 network.

Client-server architectures are addressing these shortcomings today, but many standards issues are yet to be resolved. Protocols such as lightweight directory access protocol for necessary policy-based directory services are only just beginning to be defined. Until this happens, it will be difficult for access-based IP telephony to provide features and billing that are equivalent to the features and billing available today with circuit-switched technologies.

Second, IP telephony has not completely solved the issues surrounding quality of service (QOS) in packet-switched networks. Digital technology-packet technology in particular-consists of bursts of data that are not continuous. In routing, the individual packets that make up a voice conversation take different paths through a network as they travel from origination to destination, which can delay them by different time intervals.

The voice packets can be corrupted or entirely lost. When any of these things happen, the digital-to-analog conversion that occurs before a human hears the information becomes corrupted, and the result is a perceived poor-quality voice connection.

The technical term for packet delay through the network is latency. There are a number of causes of latency, but it is usually a result of high traffic volumes on one or more of the networks that comprise the Internet. Consider that once both data and voice move as separate packets on the same network, each will contribute to the total volume of traffic over that network. To ensure that latency does not exceed acceptable levels requires managing both data and voice traffic from one end of the connection to the other. In public networks, this is not yet possible using IP telephony.

Excessive errors in packet transmission can cause delays. A common method of error correction is to retransmit, which can cause a double problem. In addition to the retransmitted packet arriving later than the original, more packets are added to the network, potentially contributing to even more delay. The number of lost packets that can be tolerated in a voice call is approximately 10%.

Both the feature and QOS issues of IP telephony will be resolved when the industry agrees on standards and vendors subsequently implement those standards. Throughout history, however, the development of telecommunications standards has often taken longer than desired. For example, it took 27 years and four revisions to define the simple RS-232 connector. ISDN needed almost a decade to reach agreement. And asynchronous transfer mode finally stabilized after multiple standards were issued over several years.

Although H.323 is a step in the right direction for IP telephony, the definitions for gatekeeper interfaces with SS7 and other networks are still evolving. If history is any indicator of the future, it will be three to five years before a vendor can build a complete IP telephony system to a stable standard definition.

The evolution to come Certainly, the existence of these technical issues does not mean that IP telephony will never reach dominance or that operators shouldn't implement it in the short term. To the contrary, visionary service providers can develop an IP telephony strategy that applies packet technology in parts of the network when and where it makes sense.

One solution is to creatively locate gateways in an operator's network. For example, a gateway at the subscriber's location could be part of a long-term strategic deployment, based on the evolving standards previously discussed. On the other hand, gateways at the network side of a telephony switch make sense now.

Such an architecture enables a service provider to offer cost- and quality-based service options, using the switch to choose routes through either the public network or the Internet. As the feature servers are developed and installed, service providers could add gateway functionality at either the HDT or the NIU. The result will be an integrated IP/circuit-switched telephony architecture (Figure 3).

In fact, because the NIU can be remotely provisioned, vendors will likely build gateway functionality into the NIU. This will allow operators to select whether a subscriber line is provisioned to route via the IP network or the public network without a truck roll dispatch. This flexibility will allow cable service providers to offer circuit-switched service today and voice over IP when the supporting infrastructure is in place. Without a doubt, this evolutionary journey will bring creative and unique solutions to the networks of tomorrow.

Tom Ruvarac is a Senior Marketing Manager in the Broadband Media Group at Tellabs Operations Inc., Lisle, Ill. His e-mail address is tom.ruvarac@tellabs.com. Justin J. Junkus is President of KnowledgeLink Inc., an Aurora, Ill.-based telecommunications consulting and training firm. His e-mail address is jjunkus@aol.com.