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Voice over IP Online Course

Lesson 4 - Quality of Service

A. QoS factors

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Factors that Affect Quality of Service

• Bandwidth
• Delay
• Jitter
• Lost packets
• Echo

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Bandwidth

Bandwidth refers to the amount of data that can pass over a network in a time frame, usually measured in bits per second (bps). If many services are using the network, then bandwidth must be shared among them. If there is not enough bandwidth, then “somethings gotta give.” Either packets can’t get onto the network in a reasonable time or excess packets are discarded. Bandwidth is listed as a contributing cause of quality of service problems because it affects the other contributing factors listed below: delay, jitter and packet loss.

Delay

Delay refers to the packets with voice data showing up fast enough that there are no gaps in the conversation.

The typical delay for a signal going end to end on the PSTN is 30 to 50 ms. Consider this the best case scenario. At the other end, delays of 250 ms or longer are intolerable. This is the typical delay when making telephone calls using satellites in geosynchronous orbit. At this delay, a caller doesn’t receive responses fast enough and speaks over the other caller. Confusion results.

End to End Signal Delay and voice quality
0 - 50 ms	Excellent quality, goal to strive for
50 - 150 ms	Acceptable voice quality
150 - 250 ms	Voice has noticeable degradation
> 250 ms	Unacceptable

Delay can be caused by many factors in a VoIP system including efficiency of the codec, processing of the packets, router delays, number of routers in the path and congestion caused by insufficient bandwidth.

Typical delay introduced by common codecs
Codec		Throughput	Typical delay
G.726	ADCPM (adaptive differential pulse-code modulation)	16, 24, 32, 40 kbps	.125 microsecond
G.728	CELP (LD-code excited linear prediction)	16 kbps	2.5 ms
G.729	CS-ACELP	8 kbps	10 ms
G.723.1	Multirate coder	5.3, 6.3 kbps	30 ms

Jitter

Jitter is defined as variable delay. In other words, the packets are showing up with different intervals between them. Jitter degrades the voice quality which becomes noticeable to the listener.

Jitter is caused by differences in packet processing, router forwarding and congestion of the network.

Lost packets

Lost packets are a fact of life on any network and an IP network is considered “best efforts”. TCP compensates for lost packets with a retransmission function. However, with real time applications such as voice and video, re-transmissions cause unacceptable delay. Therefore VoIP used UDP instead of TCP. UDP does not retransmit and therefore lost packets stay lost.

Echo

Echo in a telephone circuit refers to the speaker’s voice bouncing back from certain disjunctions of the circuit such that the speaker can hear parts of his conversation. Echo occurs even in a traditional switched telephone circuit, but since the round trip time is less than 50 ms, the effect is masked and not noticeable. When the round trip is longer than 50 ms, such as in a long distance call, echo canceling techniques need to be used.

With VoIP, round trip times are always greater than 50 ms and therefore Echo cancellation needs to be employed.

The following table lists different applications that run over a network, their typical bandwidth and their sensitivity to delay, jitter and loss. Although VoIP doesn’t take much bandwidth, it is very sensitive to problems on a network.

Applications and Quality of Service Demands
Application	Bandwidth	Sensitivity to:
		Delay	Jitter	Loss
VoIP	Low	High	High	Med
Video Conferencing	High	High	High	Med
Streaming Video on Demand	High	Med	Med	Med
Streaming Audio	Low	Med	Med	Med
Client/Server Transactions	Med	Med	Low	High
E-mail	Low	Low	Low	High
File transfer	Med	Low	Low	High

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