Fig. 4. Effects of transcoding.
merit an effort to use bandwidth efficiently. This effort
starts with the use of speech compression codecs. Use of
low bandwidth leads to a long packetization delay and
the most complex codecs. An engineering tradeoff must
be made to achieve an acceptable packetization delay, an
acceptable level of codec complexity, and an acceptable call
transmission capacity requirement. Another technique for
increasing bandwidth efficiency is voice activity detection
and silence suppression. Voice quality can be maintained
while using silence suppression if the receiving codec in-
serts a carefully designed comfort noise during each silence
period. For example, Annex B of ITU-T Recommendation
G.729 defines a robust voice activity detector that measures
the changes over time of the background noise and sends,
at a low rate, enough information to the receiver to generate
comfort noise that has the perceptual characteristics of the
background noise at the sending telephone [3].
Coding and packetization result in delays greater than
users typically experience in terrestrial switched circuit
networks. As we have seen, standard speech codecs are
available for output coding rates in the approximate range
of 64 to 5 kb/s. Generally, the lower the output rate, the
more complex the codec. Packet design involves a tradeoff
between payload efficiency (payload/total packet size) and
packetization delay (the time required to fill the packet).
For IPv4, the RTP/UDP/IP header is 40 bytes. A payload
of 40 bytes would mean 50% payload efficiency. At 64
kb/s, it only takes 5 ms to accumulate 40 bytes, but at 8
kb/s it takes 40 ms to accumulate 40 bytes. A packetization
delay of 40 ms is significant, and many VoIP systems use
20-ms packets despite the low payload efficiency when
using low-bit-rate codecs. For continuous speech, the call
transmission capacity requirement
(in kb/s) is related
to the header size
(in bits), the codec output rate (in
kb/s) and the payload sample size
(in milliseconds) as
Fig. 5 shows a plot of versus and assuming
b.
There are several header compression algorithms that
will improve payload efficiency [4]–[6]. The 40-byte
RTP/UDP/IP header can be compressed to 2–7 bytes. A typ-
ical compressed header is four bytes, including a two-byte
checksum. In an IP network, header compression must be
done on a link-by-link basis, because the header must be
restored before a router can choose an outgoing interface.
Therefore, this technique is most suitable for low-speed
access links. Fig. 6 shows a plot of
versus and
assuming b.
The lowest BW requirements lead to a long packetization
delay and the most complex codecs. An engineering tradeoff
must be made to achieve an acceptable packetization delay,
an acceptable codec complexity, and an acceptable call band-
width requirement. The following sections discuss quality
and bandwidth efficiency in more detail.
GOODE: VOICE OVER INTERNET PROTOCOL (VoIP) 1499
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