Side_1_360

The following issues have an impact on the QoS
obtained by the receiver; delay/latency, jitter,
digital sampling, voice compression, digital-to-
analogue conversion, tandem encoding, voice
activity detection, echo, packet loss and the pro-
tocols chosen. Even though the received quality
is subjective ITU have tried to make some stan-
dardized measures and from these it could be
concluded that the three main factors of the
impact of the received quality are latency, jitter
and packet loss. For latency the two main prob-
lems are echo and talker overlap. Echo becomes
a problem when the delay increases above about
50 ms, and the talker overlap becomes signifi-
cant and quite annoying when the delay is 500
ms and more. Both figures apply to the round-
trip-delay. The ITU specification G.114 [1] rec-
ommends that no more than 300 ms should
occur for the connection to be categorized as a
high quality connection. (It should be mentioned
that the one-way end-to-end delay for transmis-
sion to a satellite is approximately 270 ms.
Satellite links have been very common and are
still widely used for telephone conversations.)
For most cases the limit for echo to become
annoying will be reached and therefore VoIP
have to implement some kind of echo cancella-
tion procedures.

The delay is characterized as the total amount
of time it takes to transfer the voice from the
sender to the receiver. This time has mainly
three components; it is the propagation delay,
the serialization delay and the processing delay.
A description of the various components of the
delay is summarized in Figure 2, where all num-
bers indicate time in milliseconds. The figures
come from a study performed by Bell [2]. The
consumer objective (100 ms) and the business
objective (300 ms) are assumed values, which
are allocated to the six delay components. The
PSTN and the telephone client are neither
assumed to contribute to the total delay and are
therefore combined in the figure and marked
“negligible”. The column marked “today” is data
from actual measurements performed. The PC
Client and Gateway/POP values in the theoreti-
cal column may need some explanation. It is

assumed that two frames, with an encoding

delay of 30 ms each, are captured in one IP

packet, and that a look-ahead delay of 7.5 ms

is needed.

Packetized Voice

One of the problems of using real-time applica-

tions over packet switched networks is that

the timing of the sent packets never could be

obtained at the receiver end. This is due to the

fact that the IP networks are not isochronous, i.e.

the exact distance between two packets sent out

by the sender will most probably be changed dur-

ing the time they traverse the networks, since

they will experience different conditions on their

way to their destination. The conditions within

the network and along a route could be modelled

according to a probabilistic distribution to be able

to determine the expected delay variations. The

jitter is usually solved with a buffer; see Figure 3,

where the buffer has to be filled with Lbuffer

places before any packet is released to the

receiver. In this way, there will hopefully always

be a packet to release even though the arrival rate

of new packets has decreased momentarily. The

arrival rate to the buffer has a probabilistic distri-

bution (λ) while the release from the buffer is

done according to a deterministic distribution (d).

The packets traversing the network are not guar-

anteed in any way to reach their destination, for

datagram services not even to reach the destina-

tion in order. Individual packets could be lost

due to congestion on the links traversed. In nor-

mal cases the packet switched networks using IP

have retransmission algorithms implemented on

higher layers, i.e. the TCP on the transport layer.

The retransmission procedures are unfortunately

Figure 2 VoIP round-trip

delay allocation and current

performance in milliseconds

(ms), for the different delay

components

Figure 3 A buffer to com-

pensate for the jitter intro-

duced by the IP networks.

L is the number of packets

required in the buffer to start

releasing packets

Delay

component

PC client

Access

IP network

Gateway/POP

PSTN/phone

Total

Consumer

(objective)

100

70

50

80

Negligible

300

Business

(objective)

30

10

30

30

Negligible

100

Today

(actual)

150

150

96

160

Negligible

556

Theoretical

(minimum)

67.5

44

40

67.5

Negligible

159

Above

minimum

82,5

106

56

92,5

0

337

L

λ d