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Cisco AVVID Network Infrastructure Enterprise Quality of Service Design
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Chapter 1 Overview
Why is Quality of Service Required for AVVID?
Voice quality is directly affected by all three QoS quality factors: loss, delay, and delay variation.
Loss causes voice clipping and skips. The industry standard codec algorithms used in Cisco Digital
Signal Processor (DSP) can correct for up to 30 ms of lost voice. Cisco VoIP technology uses 20 ms
samples of voice payload per VoIP packet. Therefore, for the codec correction algorithms to be effective,
only a single Real Time Transport (RTP) packet can be lost during any given time. If two successive
voice packets are lost, the 30ms correctable window is exceeded and voice quality begins to degrade.
Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice delay becomes
too long (for example, 250 ms), the conversation begins to sound like two parties talking on over a
satellite link or even a CB radio. The ITU standard for VoIP (G.114) states that a 150 ms one-way delay
budget is acceptable for high voice quality. The Cisco Technical Marketing Team has shown that there
is a negligible difference in voice quality scores using networks built with 200 ms delay budgets.
With respect to delay variation, there are adaptive jitter buffers within Cisco IP Telephony devices.
However, these can usually only compensate for 20 to 50 ms of jitter.
Implementing QoS is a means to use bandwidth efficiently, but not a blanket substitute for bandwidth
itself. When an enterprise is faced with ever increasing congestion, a certain point is reached where QoS
alone will not solve bandwidth requirements. At such a point, nothing short of additional bandwidth will
suffice. The following sections provide guidelines that can help you determine when this point is
reached.
Voice Traffic
The bandwidth consumed by VoIP streams is calculated by adding the packet payload and all headers
(in bits), then multiplying by the packet rate per second (default of 50 packets per second).
Table 1-1
details the bandwidth per VoIP flow at a default packet rate of 50 packets per second (pps) and at 33 pps.
This does not include Layer 2 overhead and does not take into account any possible compression
schemes, such as compressed Real-time Transport Protocol (cRTP). The Service Parameters menu in
Cisco CallManager Administration can be used to adjust the packet rate.
Note
Although it is possible to configure the sampling rate above 30 ms, this usually results in very poor voice
quality.
Table 1-1
Voice Bandwidth (without Layer 2 overhead)
Bandwidth Consumption
Sampling Rate
Voice Payload
in Bytes
Packets per
Second
Bandwidth per
Conversation
G.711
20 ms
160
50
80 kbps
G.711
30 ms
240
33
74 kbps
G.729A
20 ms
20
50
24 kbps
G.729A
30 ms
30
33
19 kbps