/mcpQuality of Service for Voice over IP QoS for VoIP Overview 2 QoSVoIP.mif 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 with basic telephone services—meaning consistently high-quality voice transmissions. Like other real-time applications, VoIP is extremely bandwidth- and delay-sensitive. For VoIP transmissions to be intelligible to the receiver, voice packets should not be dropped, excessively delayed, or suffer varying delay (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-end delay for high-quality real-time traffic such as voice. (For international calls, one-way delay up to 300 ms is acceptable, especially for satellite transmission. This one-way delay takes propagation delay 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 are usually 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 and audio channel, are given priority over other kinds of network traffic. For VoIP to be deployed so that users receive an acceptable level of voice quality, VoIP traffic must be guaranteed certain compensating bandwidth, latency, and jitter requirements. QoS ensures that VoIP voice packets receive the preferential 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 to VoIP. Sufficient Bandwidth Before you consider applying any of the QoS features discussed in this document, you must first provision sufficient network bandwidth to support real-time voice traffic. For example, an 80-kbps G.711 VoIP call (64 kbps payload plus 16-kbps header) will be poor over a 64-kbps link because at least 16 kbps of the packets (which is 20 percent) will be dropped. This example also assumes that no other traffic is flowing over the link. After you provision sufficient bandwidth for voice traffic, you can take further steps to guarantee that voice packets have a certain percentage of the total bandwidth and get priority.