Chapter 3.3.6 - IP Telephony or Voice over IP

3.3.6 IP Telephony or Voice over IP

One of the paramount rules in traditional wired telephony is that service must be of
the highest quality and always available, even under disaster conditions (during a
hurricane, during electric power outage), or when the telephone device drops the
floor. Therefore, IP Telephony needs to at minimum provide the same (or almost
the same) quality with acceptable availability. It is questionable whether IP Telephony
will match the availability of traditional wired telephony; its unavailability,
however, may be compensated for by wireless telephony and other wireless communications
devices.

One of the early issues in IP Telephony was that it relies largely on digital signal
processing (DSP) coding algorithms, as compared with alpha- or mu-law
CODECs in traditional telephony, and at their first appearance, DSP coding algorithms
were not standardized, impeding the IP Telephony progress due to lack of
interoperability. Thus, the ITU-T H.323 specification came to include several
ITU-T G.700 series recommendations. For example, it includes G.723.1, which
supports two voice codec rates, 5.3 and 6.4 Kbit/s, and G.729A, which supports 8
and 13 kbit/s. It also includes other codec recommendations such as G.711, G.728,
G.729, and G.729B, as well as recommendations on video. Now, from an interoperability
point of view, compare the IP Telephony multicodec algorithms with the
traditional single-codec algorithm mu-law in the United States and alpha-law in
Europe and elsewhere. What this really means is that in IP Telephony, the packet
must include in the overhead a code to indicate to the receiving end which coding
(compression) algorithm has been used so that the decoder can adapt to this algorithm
and decompress the voice packet that has been encapsulated in the IP.
Fortunately, DSPs are programmable devices (CODECs are fixed integrated circuits)
that are able to accomplish this efficiently and at low cost (as opposed to
traditional methods).

Thus, IP Telephony implies that the Internet network must be telephony-aware;
that is, it exhibits a technological sophistication by which:

Overall delays and network latency are minimized to acceptable levels, comparable
to the traditional network.
Lost packets should be compensated for and the rate of packet loss should remain
below a threshold level commensurate with speech sample loss or burst
error rate.
The service must be reliable under extreme conditions, network and environmental,
complying to the quality of service (QoS) subscribed to.
The network should be managed to comply with the American National Standards
Institute, ANSI.1, with signaling network-management protocol
(SNMP) V1 syntax and other real-time protocols (RTP).
It addresses synchronization, clock, as well as jitter issues. The traditional
network operates on a clock that is derived from an 8 kHz global master clock
of extreme accuracy. Routers operate on their own local clock.
It addresses echo cancellation. Echo is caused by signal reflections back to
the speaker’s ear. It becomes a significant problem when overall (round-trip)
echo exceeds 50 msec.
It addresses talker overlap. This results when one-way delay exceeds 250
msecs and when one talker speaks at the same time as the other.
Contains voice compression algorithms with almost zero conversion delay to
optimize bandwidth resource utilization.
It forms “short” packets with almost zero insertion delay (the amount of
buffer delay required to form packets).
Contains call initiation and termination procedures, emulating call admission
control or traditional telephony signaling such as off-hook, on-hook, ringing,
busy, and so on.
An internet telephone may be required, which must be operable under extreme
conditions.

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