Content

| 7.5 |  | Voice Quality Issues |
| 7.5.1 |  | Common voice issues |

Two major factors affect voice quality: lost packets and delayed packets. Packet loss causes voice clipping and skips. Packet delay can cause either voice quality degradation, due to the end-to-end voice latency, or packet loss, if the delay is variable. If the end-to-end voice latency becomes too long (250 Msec, for example), the conversation begins to sound like two parties talking on a CB radio. If the delay is variable, there is a risk of jitter buffer overruns at the receiving end.

Delay
One of the most important design considerations in implementing voice is minimizing one-way, end-to-end delay. Voice traffic is real-time traffic; if there is too long a delay in voice packet delivery, speech becomes unrecognizable. Delay is inherent in voice networking and is caused by a number of different factors. An acceptable delay is less than 200 milliseconds.

There are two kinds of delays inherent in today's telephony networks: propagation delay and handling delay.

Propagation delay is caused by the characteristics of the speed of light traveling via a fiber-optic-based or copper-based medium. Handling delay (sometimes called serialization delay) is caused by the devices that handle voice information. Handling delays significantly degrade voice quality in a packet network. Delays caused by codecs are considered handling delays.

Another handling delay is the time it takes to generate a voice packet. In VoIP, the DSP generates a frame every 10 milliseconds. Two of these frames are then placed within one voice packet, so the packet delay is 20 milliseconds.

A third source of handling delay is the time it takes to move the packet to the output queue. Cisco IOS software expedites the process of determining packet destination and getting the packet to the output queue. The actual delay at the output queue is another source of handling delay and should be kept under 10 milliseconds whenever possible, by using queuing methods that are optimal for the network.

Jitter
Jitter is another factor that affects delay. Jitter is the variation of the time a voice packet is expected to be received and when it actually is received, causing discontinuity in the real-time voice stream. Voice devices compensate for jitter by setting up a playout buffer to play back voice in a smooth fashion.

End-to-End Delay
Figuring out the end-to-end delay is not difficult if the end-to-end signal paths/data paths, the codec, and the payload size of the packets are known factors. Adding the delays from the endpoints to the codecs at each end, the encoder delay (which is 5 milliseconds for the G.711 and G.726 codecs and 10 milliseconds for the G.729 codec), the packet delay, and the fixed portion of the network delay yields the end-to-end delay for the connection.

Echo
Echo is voice feedback in the telephone receiver when talking. When timed properly, echo is reassuring to the speaker. But if the echo exceeds approximately 25 milliseconds, it can be distracting and cause breaks in the conversation.

In a traditional telephony network, echo is normally caused by a mismatch in impedance from the four-wire network switch conversion to the two-wire local loop and is controlled by echo cancellers. In voice-packet-based networks, echo cancellers are built into the low bit-rate codecs and are operated on each DSP. Echo cancellers are, by design, limited by the total amount of time they will wait for the reflected speech to be received. This amount of time is called an echo trail. The echo trail is normally 32 milliseconds. VoIP has configurable echo trails of 8, 16, 24, and 32 milliseconds.

Voice, as a class of IP network traffic, has strict requirements concerning packet loss, delay, and delay variation (also known as jitter). To meet these requirements for voice traffic, the Cisco IP Telephony solution includes QoS features such as classification, queuing, traffic shaping, Compressed Real-Time Transport Protocol (CRTP), and TCP header compression. The QoS components of the Cisco AVVID IP Telephony solution are provided through the rich IP traffic management, queuing, and shaping capabilities of the Cisco AVVID network infrastructure.