This document discusses key quality of service (QoS) metrics for real-time networks including packetization, channel coding, interleaving, quantization noise, delay, jitter, and their effects. It explains sources of delay like encoding, packetization, error correction, and network queues and propagation. Interleaving rearranges bits to distribute errors while coding adds redundancy. Jitter buffers compensate for variable delay but introduce latency if too small.

1. QoS metrics

2. Packetization
• A bitstream from a network node is packetized
before it is transmitted.
• In ATM, when AAL 1 is being used, we have 5
byte ATM header, 1 byte AAL 1 header and 47
bytes to carry the data.
• Here we are assuming a constant bit rate for
the voice bitstream being carried over ATM.
• The overhead in ATM AAL1 to carry voice is 6
packets from 53 – 11.3%

3. Packetization
• In comparison, using UDP over IPv4, we get 28
bytes out of 75 being used for overhead (i.e.
37.3%) and over IPv6, we get 56 bytes
overhead out of 103 (i.e. 54.3%). Here we are
assuming 47 bytes of data.

4. Channel coding
• Done after packetization.
• Includes both interleaving and error
correction.
• Interleaving does not add extra bits – it
reorganizes the bits in the stream.
• Error correction adds bits – because it
depends on redundancy.
• Both interleaving and error correction add
delays.

5. Interleaving
• In general, to provide interleaving in blocks of
N bits, N blocks of N bits need to be buffered
in memory – the buffer size has to be N2 bits.
This is the case for the transmitter and the
receiver. This is the source of delays.
• Useful because receivers work well with
random errors but do not work well with
bursts of errors – Interleaving distributes the
errors over the bitstream.

6. Interleaving and forward error
correction
• The interleaver reads in the bits into an NxN
matrix row by row and then reads them out
column by column.
• Forward error correction comes after
interleaving and is used to protect the stream
of bits from expected errors. The simplest way
is to send M bits for every actual bit and use
the mean value of the M bits received as an
indicator of whether the source bit was 1 or 0.

7. QoS for real time networks
• The factors effecting the quality of real time
services over real time packet networks include:
–
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–
–
–
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Quantization noise.
Bit error ratio.
Delay.
Delay variation – “jitter”
Codec choice.
Echo cancellation (for voice).
Network design.
Blocking probability.

8. Quantization noise
• Due to converting an analog signal to a digital
signal.
• This is not the same as sampling noise – we
assume that the signal is correctly sampled.
• Quantization approximates the actual signal
value with a digital number.
• Because of this approximation, we get
quantization noise.

9. Quantization noise
• We can hear or see the effects of quantization
noise when we convert back to the analog
domain for playback (audio or video).
• An interpolator is used to produce a
continuous signal from the discrete signal, the
accuracy of which depends on the number of
bits that were used to approximate the
original analog signal.

10. Delay
• One way delay and two way delay.
• Delay is the time between the production of
the source signal until it is played out at the
end.
• Sources of delay:
– Source coding.
• A/D and D/A conversion.
• Frame packing.
– Packetization

11. Delay
– Channel coding
• Error detection and correction
• Interleaving
– Jitter buffer delay
– Packet queuing delay
– Propagation delay (transport delay).
• Packet queuing and propagation delays are
due to the network, the rest are due to the
end pint.

12. Frame delay
• Is the same as the frame duration. For basic
digital voice (PCM), 8000 samples are taken per
second and it is recommended that the frame
length is 1 sample – so the frame length is 0.125
ms. This is the frame delay.
• One PCM frame has 8 bits (1 byte for each
sample).
• Code Excited Linear Prediction (CELP) based
codecs typically have a frame delay from 5 to 10
ms.

13. Packetization delay
• Incurred because the packetizer needs to wait for
the payload bits prior to completing the packet
header.
• For ATM, the packetizer needs to wait for 47
bytes prior to sending the full ATM cell.
• If the bits are being produced at 64 kbps, then 47
bytes need 5.88 ms to be collected. The same
delay is incurred at the receiver, meaning that
there is a total delay of 11.76 ms.

14. Interleaving delay
• Interleaving delay can be calculated by:
– Interleaving block size = (n x m x l) bits
– Interleaving buffer size = (n x m x l)2 bits
– Delay = buffer size / r (in ms)
• n: number of speech frames interleaved.
• m: number of speech samples contained in one speech
frame.
• l: number of bits per sample
• r: codec bit rate in kbps

15. Error correction and jitter buffer delay
• Depends on the correction scheme used.
Better correction schemes (i.e. higher
protection) have a higher delay typically.
• Jitter (delay variations) is typically handled by
buffered packets and playing them out at a
constant rate. This buffering introduces a
delay. Typically, 60 ms of real time data is
buffered.

16. Queuing delay and propagation delay
• These are delays from the network and not
the source.
• Queuing delay is due to the packet being
queued at different points in the network
before being “moved on”. E.g. being queued
at a router before being processed.
– As long as the arrival rate of the packets is lower
than the processing rate, then a buffer can be
used to ensure no packet loss. Otherwise packet
loss will occur.

17. Queuing and propagation delay
• Propagation delay is the “speed of light delay”.
– In a vacuum, the speed of light is 3x108 m/s.
– Across the US, the microwave transmission distance is
3,676 km. This requires 12.25 ms to be traversed.
– Another example, geostationary satellites. These are
at 29961 km from the Earth’s surface.
– For these satellites, 100 to 121 ms are required
between the Earth and the satellite. End to end
sattellite communications requires upto 242 ms, if no
intersatellite delay is included.

18. Effect of delay
• Increased delay means that it becomes more
difficult to deal with echo (to cancel the echo).
• Communicating with echo is very annoying
and can make a conversation near impossible.
• The ITU-T recommends (in G.114):
– 0-150 ms delay for local connections.
– 150-400 ms for international connections.
– 400ms and above is unacceptable for general
networking purposes.

19. Delay variation (Jitter)
• Jitter occurs because different packets arrive
at different times.
• A jitter buffer is used to smooth the playout.
• If the delay variation is greater than the jitter
buffer, then packets will be lost.
• Jitter is typically due to packet queuing, since
the other sources of delay are constant.
• By reducing the arrival rate, jitter can be
reduced dramatically.