Quality of Service for Voice over IPQoS for VoIP Overview
QoS for VoIP Overview
For VoIP to be a realistic replacement for standard public switched telephone network (PSTN)telephony services, customers need to receive the same quality of voice transmission they receive withbasic telephone services—meaning consistently high-quality voice transmissions. Like other real-timeapplications, VoIP is extremely bandwidth- and delay-sensitive. For VoIP transmissions to beintelligible to the receiver, voice packets should not be dropped, excessively delayed, or suffer varyingdelay (otherwise known as jitter). For example, the following standards must be met:
The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors. Ideally,there should be no packet loss for VoIP.
The ITU G.114 specification recommends less than 150 millisecond (ms) one-way end-to-enddelay for high-quality real-time traffic such as voice. (For international calls, one-way delay up to300 ms is acceptable, especially for satellite transmission. This one-way delay takes propagationdelay into consideration—the time required for the signal to travel the distance.)
Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and areusually only effective on delay variations less than 100 ms. Jitter must therefore be minimized.VoIP can guarantee high-quality voice transmission only if the voice packets, for both the signaling andaudio channel, are given priority over other kinds of network traffic. For VoIP to be deployed so thatusers receive an acceptable level of voice quality, VoIP traffic must be guaranteed certain compensatingbandwidth, latency, and jitter requirements. QoS ensures that VoIP voice packets receive thepreferential treatment they require. In general, QoS provides better (and more predictable) network service by providing the following features:
Supporting dedicated bandwidth
Improving loss characteristics
Avoiding and managing network congestion
Shaping network traffic
Setting traffic priorities across the network
Quality of Service for Voice over IP
discusses various QoS concepts and features that are applicable toVoIP.
Before you consider applying any of the QoS features discussed in this document, you must firstprovision sufficient network bandwidth to support real-time voice traffic. For example, an 80-kbpsG.711 VoIP call (64 kbps payload plus 16-kbps header) will be poor over a 64-kbps link because at least16 kbps of the packets (which is 20 percent) will be dropped. This example also assumes that no othertraffic is flowing over the link. After you provision sufficient bandwidth for voice traffic, you can takefurther steps to guarantee that voice packets have a certain percentage of the total bandwidth and getpriority.