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Factors Affecting QoS
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The four most important network parameters for the effective transport of VoIP traffic over a softswitched network are bandwidth, delay, jitter, echo, and packet loss (see Table 8-1). Voice quality is a highly subjective thing to measure. This presents a challenge for network designers who must first focus on these issues in order to deliver the best voice quality possible. Using voice quality as a performance metric for network configurations provides appropriate mechanisms for obtaining and correlating both subjective and objective data. Perhaps the simplest test of a VoIP network is to call your mother. That is, the simplest metric of QoS of a VoIP network is in the ear of the beholder. Just as reliability (achieving five 9s) in a network is simply a matter of good engineering, so is QoS. This chapter will explore the solutions available to service providers that will deliver the best QoS possible.
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Quality of Service (QoS)
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Quality of Service (QoS)
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Table 8-1 Factors affecting QoS
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Description The time from transmission of a packet to its reception. This is measured in units of time. For VoIP transmissions that is usually a matter of milliseconds. The variation in arrival times between continuous packets transmitted from point A to point B. This is caused by packet-routing changes, congestion, and processing delays. Greater bandwidth delivers better voice quality. The percentage of packets never received at the destination.
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Factor Delay
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Bandwidth Packet Loss
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Just as it is necessary to closely examine every element of the network for anything that might detract from reliability when planning to deliver High Availability (HA), it is necessary to scrutinize the network for any element that might induce delay, jitter, packet loss, or echo. This includes first the hardware elements, such as router and media gateways, and second the routing protocols that prioritize voice packets over all other types of traffic on the IP network.
Improving QoS in IP Routers and the Gateway
End-to-end delay is the time required for a signal generated at the talker to reach the listener. Delay is the impairment that receives the most attention in the media gateway industry. It can be corrected via functions contained in the IP network routers, the VoIP gateway, and in engineering in the IP network. The shorter the end-to-end delay, the better the perceived quality and overall user experience. The following sections discuss sources of delay.
Sources of Delay: IP Routers
Packet delay is primarily determined by the buffering, queuing, and switching or routing delay of the IP routers. Packet capture delay is the time required to receive the entire packet before processing and forwarding it
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Quality of Service (QoS)
8
through the router. This delay is determined by the packet length, link-layer operating parameters, and transmission speed. Using short packets over high-speed trunks can easily shorten the delay. VoIP networks use packetization rates to balance connection bandwidth efficiency and packet delay. Switching or routing delay is the time it takes a network element to forward a packet. New IP switches can significantly speed up the routing process by making routing decisions and forwarding the traffic in hardware devices instead of software. Due to the statistical multiplexing nature of IP networks and the asynchronous nature of packet arrivals, some delay in queuing is required at input and output ports of a packet switch. Overprovisioning router and link capacities can reduce this delay in queuing time.
Sources of Delay: VoIP Gateways
Voice-signal processing at the sending and receiving ends adds to the delay and includes the time required to encode or decode the voice signal from analog or digital form into the voice-coding scheme selected for the call and vice versa. Compressing the voice signal also increases the delay. The greater the compression, the greater the delay. When bandwidth costs are not a concern, a service provider can utilize G.711, which is uncompressed voice, and this imposes a minimum of delay due to the lack of compression. Later in the chapter the parameters for G.711 and other voice-coding standards will be discussed. On the transmit side, packetization delay is another factor that must be entered into the calculations. The packetization delay is the time it takes to fill a packet with data. The larger the packet size, the more time is required. Using shorter packet sizes can shorten this delay but will increase the overhead because more packets have to be sent, all containing similar information in the header. Balancing voice quality, packetization delay, and bandwidth utilization efficiency is very important to the service provider.1 How much delay is too much Of all the factors discussed in 4, Voice over Internet Protocol, that degrade VoIP, latency, or delay, is the greatest. Recent testing by Mier Labs offers a metric as to how much latency is acceptable or comparable to toll quality, the voice quality offered by the PSTN. Latency less than 100 milliseconds does not affect toll-quality voice. However, latency over 120 milliseconds is discernable to most callers,
Douskalis, Bill. IP Telephony: The Integration of Robust VoIP Services. New York: McGraw-Hill, 2000, pg. 230 231.
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Quality of Service (QoS)
Quality of Service (QoS)
and at 150 milliseconds the voice quality is noticeably impaired, resulting in less than a toll-quality communication. The challenge for VoIP service providers and their vendors is to get the latency of any conversation on their network to not exceed 100 milliseconds.2 Humans are intolerant of speech delays of more than about 200 milliseconds. As mentioned earlier, ITU-T G.114 specifies that delay is not to exceed 150 milliseconds one way or 300 milliseconds round trip. The dilemma is that although elastic applications (email for example) can tolerate a fair amount of delay, they usually try to consume every bit of network capacity they can. In contrast, voice applications need only small amounts of the network, but that amount has to be available immediately (see Figure 8-1).3
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