INTRODUCTION TO VOICE OVER IP

52-30-20 DATA COMMUNICATIONS MANAGEMENT INTRODUCTION TO VOICE OVER IP Gilbert Held INSIDE Equipment Utilization; VoIP Gateway; Router with Voice Modules; IP Gateway; Latency; Delay Components; Encoding; Local LAN Delay; Local Router Processing Delay; Local Access Line Delay; Network Delay; Remote Access Line Delay; Remote Router Delay; Remote LAN Delay; Remote Voice Digitization-Decoding OVERVIEW This article focuses attention on the methods that organizations can use to implement VoIP. In doing so, it examines the use of voice-enabled routers and gateways as well as the key metric that governs a successful implementation of VoIP latency. In examining the major components of latency, note that several techniques can be used to adjust delay through a network. In addition, also note why VoIP in a business environment is currently suitable for a private network rather than for use via the Internet. EQUIPMENT UTILIZATION Today there are two main equipment utilization strategies that businesses are considering for implementing VoIP. One strategy involves the use of gateways attached to a PBX and a LAN. As a participant on a LAN, the gateway must use the services of a separate router. A second equipment PAYOFF IDEA Over the past two years, one of the hottest technologies in the field of communications has been Voice over the Internet Protocol, commonly referred to as Voice over IP, as well as by the mnemonic VoIP. The reason for the key interest in VoIP is the economics associated with the technology. For organizations with an existing TCP/IP network infrastructure, it becomes possible to add a voice transmission capability to a network for a penny or less per minute. Thinking back to a few years ago, one might remember Candice Bergman, Sprint Corporation s Dime Lady, extolling the virtue of signing up for that communication carrier s dime-per-minute plan for calling during the evening and on weekends. Because it is now possible for businesses to use their existing network infrastructure for voice calling at any time during the day for a tenth of the Sprint restrictive calling rate, it is easy to visualize the interest in the technology. Thus, the major reason for organizations considering the implementation of VoIP is one of economics. However, there are also other reasons for considering VoIP. Those other reasons include the ability to integrate voice and data networks, which can reduce communications personnel requirements, and the ability to consolidate the floor space and power required to operate the equipment. Auerbach Publications 2001 CRC Press LLC

DATA COMMUNICATIONS MANAGEMENT utilization strategy involves adding voice modules to a router. In this equipment environment, the voice modules are connected to a PBX and the router performs the required routing, eliminating the need for a separate router. A third equipment utilization strategy that warrants mention is an IP PBX. Here, the idea behind the IP PBX is to replace a conventional PBX by a computer-based platform that performs voice digitization and packetizes voice samples into IP packets for routing through the computer. Because an IP PBX uses a VoIP gateway at its core, brief mention is made of some additional equipment used to form this new type of PBX. However, it should also be noted that at the present time this new type of PBX typically lacks several functions that are included on most conventional PBXs. Such functions include three-way calling, follow-me calling, call forwarding, music on hold, speed dialing, and similar features. In addition, IP PBXs typically do not scale anywhere near the level of large-scale conventional PBXs that can support up to tens of thousands of lines. Due to the preceding, the first generation of IP PBXs has not obtained more than a token level of acceptance and thus is only briefly discussed in this article. The top portion of Exhibit 1 illustrates the use of a VoIP gateway in an office environment. The middle portion of Exhibit 1 illustrates the use of a router with voice modules to provide a VoIP capability. Finally, the lower portion of Exhibit 1 illustrates an example of an IP PBX configuration. VoIP GATEWAY In examining the three equipment configurations shown in Exhibit 1, proceed in a top-down approach commensurate with the order of devices in the illustration. In examining the configuration for the VoIP gateway, note that it is a participant on the LAN. This means that the level of LAN utilization will adversely affect delay or latency of packets transporting voice. Also note that there are many types of VoIP gateways. Some gateways have digital interfaces, permitting T1 lines that support up to 24 simultaneous calls. Other VoIP gateways are limited to analog interfaces, requiring one port to support one call. As previously mentioned, the VoIP gateway solution requires the use of a separate router; however, this solution scales up to hundreds or perhaps a thousand or more ports. ROUTER WITH VOICE MODULES The second equipment configuration, shown in the middle of Exhibit 1, requires the use of a router with voice modules. In examining the use of a router with voice modules, note that although the router is a participant on the LAN, packets transporting digitized voice do not flow over the LAN. Instead, such packets flow directly from the router where they are formed onto the serial port connected to the IP network.

INTRODUCTION TO VOICE OVER IP EXHIBIT 1 Examining VoIP Equipment Configurations A. VoIP Gateway Digital PBX IP Network Public Switched Telephone System (PSTN) VoIP Gateway Router Analog PBX LAN B. Using a Router with Voice Modules Digital PBX IP Network Public Switched Telephone System (PSTN) Analog PBX Router with Voice Modules LAN C. IP PBX Call Manager Router IP Network LAN Public Switched Telephone System (PSTN) IP PBX Ethernet IP Telephone Ethernet IP Telephone Because it is possible to set router queues that provide a preference for output onto the serial port for digitized voice packets over packets received via the LAN, the use of a router with voice modules minimizes latency onto the IP network. This can be a significant issue if the level of LAN utilization is relatively high. This is because when the level of LAN utilization is high and packets transporting digitized voice must flow over a LAN, collisions can significantly delay their transport to the router for placement into the IP network. While a router with voice modules can be expected to have a lower level of latency than when a VoIP gateway is used, there are two disad-

DATA COMMUNICATIONS MANAGEMENT vantages associated with this equipment configuration. First, not all routers are upgradable to support voice modules. This means an organization may have to replace an existing router to take advantage of the lower latency of this network equipment configuration. A second disadvantage of this network equipment configuration concerns scalability. Although there are both digital and analog voice modules that can be added to a router, the actual number of modules that can be added is limited. For example, a Cisco 2600 series router only has one system expansion slot. This means one can only add one board to this router. That board can contain one digital T1 voice module providing support for up to 24 calls or up to eight analog voice modules, which now limits the router to supporting up to eight calls. IP GATEWAY The third equipment configuration shown in Exhibit 1 is that of an IP PBX. In examining the lower portion of Exhibit 1, note that while the IP BPX replaces the use of an analog or digital PBX, it is not a fully integrated device. Many vendors require a separate call manager, with both the call manager and IP PBX becoming participants on a LAN. Also note that an IP PBX typically uses the LAN to transmit packets containing digitized voice to Ethernet IP telephones. Concerning the latter, the IP Ethernet telephones may either be directly connected to a LAN or interfaced to a PC that is connected to a LAN. For either situation, a separate router is required if voice is to flow over an IP network. In addition, the transport of voice packets over the LAN will add to end-to-end latency, which will be important when such packets are routed onto the IP network for transport to a distant location. Having a general indication of three common VoIP equipment configurations that can be used to integrate voice into an IP network, one can now focus on the key issue that will determine the success or failure of the integration process. That key issue is latency. LATENCY The end-to-end latency (or delay) a packet transporting digitized voice encounters is the key issue that governs the success or failure of a VoIP implementation. The reason latency is a key issue is related to the manner by which humans converse. If a delay exceeding a quarter of a second (250 ms) occurs during a conversation, the party at each end of the conversation tends to believe that the other party is now waiting for a response. At this point, there is a high probability that both parties to the conversation will do something humans normally do not do when talking in a civilized manner. That is, both parties will tend to talk at the same time. When this occurs, the two parties will stop talking and attempt to negotiate an orderly method to converse.

INTRODUCTION TO VOICE OVER IP One might remember the use of Citizen Band (CB) radios, in which the parties to a conversation used the term over for one person to tell the other that he or she was finished talking and the other person could talk. In a business environment, it is highly doubtful that people would put up with the use of a CB-like communications method to obtain the ability to integrate voice into an IP network. Due to this, the key limiting constraint associated with integrating voice into an IP network is to obtain an end-to-end latency or delay less than 250 ms. Prior to examining the delay components associated with a VoIP network environment, a few additional words about latency are warranted. While this author considers a maximum latency of 250 ms as an acceptable delay metric, the International Telecommunications Union (ITU) has a more stringent specification. The ITU specifies that a conventional voice telephone system should not have a round-trip delay that exceeds 300 ms, or 150 ms on a one-way, end-to-end basis. However, it should be noted that the ITU specification was developed prior to packetized voice becoming a reality. While a one-way delay difference between 150 and 250 ms is significant, this author believes that for most people a latency under 250 ms will allow an acceptable level of voice communications to occur. Given an appreciation for the maximum latency a VoIP environment should have, now take a look at the components of delay and how some components can be adjusted. DELAY COMPONENTS If one considers the manner by which a voice conversation is digitized, packetized, and routed through a network to its intended destination, there are nine areas where the vast majority of delay occurs. Those delay areas are listed in Exhibit 2. EXHIBIT 2 Causes of the Majority of Delays in a Voice over IP Networking Environment Delays in a voice over IP networking environment can be attributed to the following nine areas: 1. Local voice digitization: encoding 2. Local LAN delay 3. Local router processing delay 4. Local access line delay 5. Network delay 6. Remote access line delay 7. Remote router delay 8. Remote LAN delay 9. Remote voice digitization: decoding

DATA COMMUNICATIONS MANAGEMENT Encoding With the exception of network delay, the method of voice encoding can be expected to contribute the most to end-to-end latency. The actual encoding delay is a function of the voice coding method used. In general, the higher the encoding bit rate, the lower the latency associated with the coding method. For example, PCM at 64 Kbps has a delay of less than 1 µs. In comparison, low bit rate coders, such as a 5.3-Kbps version of Code Excited Linear Prediction (CELP) has an encoding delay of 67.5 ms. Due to the significant differences in encoder delays, a VoIP application with an end-to-end latency exceeding 250 ms that could be detrimental to the hearing of users might be workable using a different encoder. Thus, prior to giving up on VoIP, one should consider the delays associated with different voice coders and the possible selection of a different coder. Local LAN Delay If equipment configuration requires the use of a LAN, packets will experience some delay in reaching the router. To determine the delay, one can use the PING utility program and ping the router s Ethernet port from a workstation on the network. Local Router Processing Delay The processing delay of a router is based on the workload presented to the router. One can determine the delay through the local router using Telnet to access the router. Then use the router s internal PING command to ping its serial port to determine the round-trip delay through the router. If this delay divided by two (2) is more than a few milliseconds, consider establishing a priority queue for packets transporting digitized voice to expedite their flow through the router onto the IP network. Local Access Line Delay The operating rate of the local access line has a bearing on end-to-end delay. For example, transmission via a 56-Kbps line into an IP network results in a delay 24 times that of a T1 line. Thus, one may wish to consider the use of different types of access lines, such as fractional T1 and T1. Network Delay The routing through an IP network normally accounts for a majority of the end-to-end delay. In a corporate IP environment, one can control delay via the creation of priority queues in each network router to expedite the flow of VoIP. In addition, one can use RSVP to guarantee bandwidth to VoIP.

INTRODUCTION TO VOICE OVER IP In an Internet environment, both of the previously mentioned methods are not at present applicable. Thus, it is far easier to tailor a private IP network for voice than to use the Internet. This explains why a majority of VoIP implementations are at present occurring on private IP networks. Remote Access Line Delay Similar to the local access line delay, the access line at the remote site will also contribute to latency. Thus, one can control remote access line delay by considering the use of different types of access lines, such as 56 Kbps, fractional T1, and T1 lines. Remote Router Delay The remote router delay is normally negligible in comparison to the delay associated with the local router. This is because the remote router receives data at the access line operating rate that is normally a fraction of the LAN operating rate. Thus, there is normally a slight delay through the remote router. However, one can still prioritize VoIP traffic by establishing router queues that expedite traffic to VoIP gateways or an IP PBX from the remote router. Remote LAN Delay Similar to the local LAN, any traffic to a VoIP gateway or IP PBX that flows over the remote LAN will experience delay based on the level of network utilization. Thus, one may wish to examine the level of remote LAN utilization as well as use the PING utility to determine the delay on the remote LAN from a workstation to the remote router. Remote Voice Digitization-Decoding The final contributor to end-to-end latency is represented by voice decoding at the remote site. Unlike voice encoding that can vary considerably based on the encode used, decoding is typically 5 to 20 ms. However, part of the decoding process depends on the setting of a jitter buffer whose improper use can literally kill a VoIP application. Thus, a few words about the use of the jitter buffer are in order. The jitter buffer is built into encoder/decoder chips and provides a mechanism to compensate for the random delays that packets encounter as they flow through an IP network. Because VoIP applications use the Real Time Protocol (RTP) to timestamp packets, it becomes possible to smooth out packet delays by placing packets into a jitter buffer and extracting them from the buffer based on their RTP timestamp. Most jitter buffers support settings from 0 to 255 ms, with 0 disabling the buffer. For some unknown reason, the default setting on some products is 255 ms, which exceeds the maximum one-way delay. For most VoIP applications,

DATA COMMUNICATIONS MANAGEMENT the jitter buffer should be set to between 10 and 20 ms, which provides sufficient time associated with the flow of packets through an IP network. SUMMARY As indicated in this article, both the configuration of equipment as well as latency play a major role in the suitability of VoIP. By examining the components contributing to latency and adjusting them when required, it may be possible to tailor one s network environment to make the integration of voice into an organization s IP network a success. Gilbert Held is an award-winning author and lecturer. Gil is the author of over 50 books and 400 technical articles. Two of his recent publications are Introduction to Data Communications, 3rd ed., published by John Wiley & Sons, and Cisco Router Performance Field Guide, published by McGraw-Hill. Gil can be reached via e-mail at gil_held@yahoo.com.