Produced by Network World Technical Seminars in association with DigiNet Corporation

Are You Ready for Voice over IP?

The following technology report is produced by DigiNet Corporation (http://www.diginet.com), in association with Network World Technical Seminars.

There is a lot of buzz these days about extending the capabilities of IP-based internetworks to transmit voice, video and multimedia information, in addition to the more familiar applications of email, Web surfing and file transfers. The initial argument in favor of a converged network states that an IP-based infrastructure is likely already in place to serve the needs of the corporate data environment. Some of this network capacity is probably going unused. So if this unused bandwidth is re-provisioned to support voice and fax traffic, then greater network utilization efficiencies result. In addition, synergies in network design, deployment and management may also occur. In effect, your voice and fax calls may be able to ride for free on your existing data network.

This initial argument is supported by business research into these new technologies, which indicates that the voice over IP (VoIP) market has a huge potential for growth. In fact, International Data Corp. has projected that between 1997 and 2002 the domestic revenues for VoIP will grow at a compound annual growth rate of 103.4%, reaching $24.39 billion in 2002. On the international side, the annual growth rate for the same period is projected at 100.9%, reaching $20.49 billion in 2002.

Given these impressive growth rates, network managers need to learn more about VoIP technologies. But is your network infrastructure a candidate for this new mix of traffic? And for that matter, is your organization prepared to design, configure and manage a multimedia internetwork?

Following are some implementation issues for your consideration:

1. Network convergence is a challenge because voice and data networks are fundamentally different.

2. The differences between voice and data network operation will require different management approaches.

3. Quality of Service factors must be considered.

4. New standards must be implemented.

5. Multi-vendor interoperability will be a key ingredient.

6. Other driving factors.

1. Network convergence is a challenge because voice and data networks are fundamentally different.

Voice and fax networks, which employ circuit switching, are characterized by the following:

• Call setup is required to initiate the connection.
  
  
• Network resources are reserved for the duration of the connection.
  
  
• A fixed bandwidth (typically 64 Kbps per voice channel) is used, and may or may not be consumed depending upon the traffic.
  
  
• Pricing is typically based upon the duration of usage.
  
  
• Providers may be regulated by a government agency, such as the Federal Communications Commission or state Public Utilities Commissions.
  
  
• Service should be ubiquitous and readily available to the masses.

In contrast, data networks, which employ packet switching, are identified by the following characteristics:

• Per-packet addressing is required to assure data delivery, with no requirement for a call setup.
  
  
• Network resources are consumed as needed, and typically not reserved on an end-to-end basis.
  
  
• Pricing is largely based upon competition, not upon regulatory edicts.
  
  
• Service is provided on-demand, and may be constrained by geographic location, speed of the technology implementation, marketplace competition, etc.

Thus, trying to shoe-horn a voice, fax or video call into a packet network may be met with some fundamental technical challenges. For starters, what happens if a packet carrying a voice or video clip is delayed or lost? How will this loss be perceived at the receiving end? Will the signal quality suffer in the mind of the (human) receiver?

Implementing a converged network means that you are willing to study the differences between voice and data network characteristics, understand the technical challenges that are then presented, and address these challenges with the end users' perspective in mind (see Figure 1).

2. The differences between voice and data network operation will require different management approaches.

The U.S. telephone industry has earned a world-wide reputation for highly reliable service. The term "five nines" of reliability, or 99.999%, is frequently quoted as the benchmark against which telephone network equipment is measured. This figure comes from some of the design objectives behind central office switching systems that are specified for a downtime of 2 hours in 40 years of operation. Forty years is approximately 350,400 hours. Two hours of downtime would translate to 0.0000057 of that time period. Translating the downtime calculation into an uptime estimate yields an uptime of 99.9994% (the "five nines").

Admittedly, many end users of the voice network may not know (or, for that matter, may not care) whether they have three, four or five "nines" of reliability in that network. Their experience tells them that call setups occur in a matter of a few seconds, calls are almost always completed to the correct destination, and call disconnects rarely occur. With this level of experienced reliability, any network failures, or quality of service degradations, will stick out like a very sore thumb. If your end users can't get a dial tone, phone home or dial up the remote access server and check email, you will likely hear about it.

In contrast, data network users may be more tolerant of disruptions that effect their data transmission. They have all experienced difficulties with their own workstations, application programs, etc. and seem to be (slightly) more understanding of server problems, router re-configurations and so on. But what happens if you move application traffic from an ultra-reliable network (such as the Public Switched Telephone Network) to a less-than-ultra-reliable network (such as the global Internet or your corporate Intranet)? Will the same reliability assumptions still hold? Will you be able to manage the converged network in the same manner? Will your previous management procedures be as effective?

If you are planning to change the voice communication infrastructure from a circuit-switched to a packet-switched environment, you must assure that the quality of service of the new system at least matches the quality of the old system. Otherwise, you may have some very unhappy constituents on your hands.

3. Quality of Service factors must be considered.

Delivering voice, fax and video signals from one point to another cannot be considered successful unless the quality of the delivered signal satisfies the recipient. There are many elements that effect this quality, including:

• Bandwidth requirements: the transmission speed of the network infrastructure and physical topology of that infrastructure.
  
  
• Control functions: including resource reservation, provision and monitoring that are required to set up and maintain the multimedia connection.
  
  
• Latency or delay: from the signal source to the signal destination through the network.
  
  
• Jitter: a variation in arrival times between packets, which requires the incoming packets to be placed in a buffer and then released from that buffer at standard intervals.
  
  
• Packet Loss: when a voice or video packet is lost in the network, thus requiring some type of signal compensation at the receiving end.

Of these factors, latency is the key element effecting quality of service. With most systems, a round-trip latency of 400 milliseconds is considered the maximum tolerable delay, with a round-trip latency of 200 milliseconds more optimal. When latency exceeds these limits, the quality of the received voice signal degrades dramatically.

Latency is comprised of several components. The propagation delay of the transmission media is approximately 1 nanosecond per foot, which accumulates to a one-way delay of 10-100 milliseconds depending upon the physical distance between the two locations in question. The signal encoding/decoding algorithms add approximately 20 milliseconds of delay at each end of the connection, depending upon the algorithm in use. Jitter buffers at the receiver may add 40 milliseconds of delay. Packet processing and queuing functions may add another 10-40 milliseconds of delay, depending upon the speed of the connection, the maximum transmission unit (MTU) size, etc.

4. New standards must be implemented.

Standards are the blueprints through which voice and data communication networks are designed, implemented and managed. A number of organizations are recognized as international standards bodies, including the American National Standards Institute (ANSI), the Institute of Electrical and Electronics Engineers (IEEE), the International Organization for Standardization (ISO), the International Telecommunication Union (ITU) and the Internet Engineering Task Force (IETF).

Voice, fax and video over IP networks touch both the voice and data aspects of communication technologies, and therefore the standards must address both of these elements. Some of the key families of standards include:

• T.120 Series: published by the International Telecommunication Union - Telecommunication Standards Sector (ITU-T), defines protocols and services for real-time, multipoint data communication. The protocols may be implemented over many different types of networking infrastructures, including circuit and packet-switched networks, Integrated Services Digital Networks (ISDNs) and Local Area Networks (LANs). Applications for this technology include data conferencing, whiteboard image sharing and still image exchange.
  
  
• H.320 Series: published by the International Telecommunication Union - Telecommunication Standards Sector (ITU-T), defines video telephony communications over point-to-point and multipoint connections. The most notable standard in this series is H.323, which specifies packet-based multimedia systems.
  
  
• G.700 Series: published by the International Telecommunication Union - Telecommunication Standards Sector (ITU-T), defines the technical aspects of terminal equipment, including the specific algorithms that are used to encode and decode analog signals (such as voice) for transmission over a digital network. An example of a standard in this series is G.723.1, which defines voice encoding at 5.3 and 6.3 Kbps.
  
  
• T.37 and T.38: published by the International Telecommunication Union - Telecommunication Standards Sector (ITU-T), defines Store- and Forward-Fax and Real-Time Fax message formats and procedures, respectively.
  
  
• Request for Comments Series: published by the Internet Engineering Task Force, these documents specify the operation of the global Internet and are used by many corporate intranets that connect to the Internet. Examples of documents that are used in voice, fax and video over IP implementations include RFC 1889 (Real Time Protocol), RFC 2068 (Hypertext Transfer Protocol), RFC 2205 (Resource Reservation Protocol), RFC 2326 (Real-time Streaming Protocol) and RFC 2327 (Session Description Protocol).
  
  
• VoIP Protocol Stack: published by the Voice over IP Forum, this document defines the specific ITU-T and IETF standards that are required in support of a multi-vendor, interoperable protocol implementation.

Understanding the importance of these standards, and knowing their key elements, enables the network manager with the technical references upon which to build a converged internetwork.

5. Multi-vendor interoperability will be a key ingredient.

Remember the days when one vendor's Ethernet card would not communicate with another vendor's card? Some of the lessons from the earlier days of networking need to be remembered as the industry moves into the era of multimedia communication. Fortunately, the industry, led by the International Multimedia Teleconferencing Consortium (IMTC), has made major strides in this critical area.

The IMTC, with over 150 member companies, is chartered to promote, develop and implement interoperable multimedia teleconferencing solutions that are based upon open international standards. In conjunction with these efforts, the IMTC sponsors periodic interoperability tests that provide a forum for member companies' engineering staffs to verify their implementations vis-a-vis their peers in other organizations. The outcome of those tests is then used to refine product development, and assure greater compatibility between different vendors' solutions.

As multimedia applications become more ubiquitous, the assurance of interoperable vendor implementations will become more important. Network managers that are venturing into this new arena should check out the IMTC's website at www.imtc.org.

6. Other driving factors.

In addition to the design and management issues noted above, there are other forces, some that are beyond your control, that will shape your migration to a converged network. For example, will the IP Telephony Service Providers (ITSPs) be subject to federal regulations? Will access charges be imposed on VoIP calls? Does the Internet have the capacity for the expected increase in IP traffic? How will the growing number of ITSPs distinguish their services from that of their competition? Will the key PBX vendors emerge as the strongest gateway suppliers, or will the router vendors mount a significant challenge from the data networking side and increase their market share in this new area? Are network management and protocol analysis vendors prepared to support this new market?

All of these questions are being asked, and network managers will demand clear and concise answers from the VoIP industry. In short, the interest in VoIP goes beyond free voice and fax calls over the Internet. Instead, it extends to the very fundamental core of how business communication will serve end users into the next millennium, and the economies of scale that a single, converged network infrastructure will provide.