A Study of the ITU-T G.729 Speech Coding Algorithm ...

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MASTER THESIS
Datum - Date Rev Dokumentnr - Document no.
04-09-28 PA1
Several methods and sophisticated algorithms have been developed for the evaluation
of the QoS. PSQM (ITU P.861), PAMS (BT) and PESQ are some examples [23].
The Resource Reservation Protocol (RSVP) is a protocol which allows the Internet to
support QoS. RSVP, which is part of the Internet Integrated Service (IIS) model, contains
the ability to reserve resources along a path from the source to the destination, which
then in turn allows routers, enabled by RSVP, to schedule and prioritize packets, which
guarantees the QoS.
IP-IPv6, which succeeded IPv4, inherently supports QoS. However, in contrast to IPv4,
whose packet header requires a mere 20 bytes, the packet header of IPv6 necessitates
40 bytes, which doubles the overhead and may thus result in a greater latency which is
especially troubling for vocoders with diminutive packets. However, header compression
schemes are existent which are able to, at least partially, offset the extended header overhead.
4.5.1 Packet Loss and Jitters
If a router receives too many packets due to network congestions, some of the packets
may be dropped out, known as packet loss. In general, packet loss has a substantial effect
on the quality of the received audio/video and packet losses in excess of 10 percent
are regarded as intolerable. Since voice transmissions are extremely time sensitive, regular
retransmission instruments which are based on TCP appear inappropriate. Speech
interpolation is one approach commonly used for packet-loss compensation. With interpolation,
losses which incur during unvoiced speech segments are corrected through packet
repetition, while losses during voiced speech are repaired through the use of pitch-cycle
waveform repetition.
Even in the case that packets are transmitted at even intervals, they may reach the recipient
with uneven intervals. Such a gap of arrival time is known as jitter. Jitters are generally
the result of network congestions, because routers have to handle a great number
of packets transmitted from many other hosts. Since jitters affect the timing of audio/video
playback, also the quality of received audio and video is decreased. The removal of jitters
requires the collection of packets, which subsequently have to be stored long enough so
that the slowest packet will arrive and be played in the correct sequence. A jitter buffer
temporarily stores arriving packets thereby minimizing delay variations. Packets that arrive
too late are discarded [21].
4.5.2 Latency
Latency is an essential factor in IP performance, and reducing or mitigating the effects
of latency is an important aspect of network performance engineering. Latency, which is
defined as the time delay incurred in speech by the IP Telephony system, is generally measured
in milliseconds from the moment that the speaker utters a sound until the listener
actually hears the sound. This is known as one-way latency, while round-trip latency is the
sum of the two one-way latency figures which make up a telephone call. The round-trip
latency for domestic calls over the PSTN within the continental United States, is, for instance,
virtually always under 150 milliseconds. At these levels, users are unable to notice
the latency. By contrast, international calls, particularly the ones carried via satellite, may
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