IP TELEPHONY: Camouflaging Advantages

The threat to circuit-switched telephone services by IP telephony gets

neutralised by a bunch of Quality of Service (QoS) issues unique to packet

networks. These issues more often than not camouflage the advantages of the

reduced cost and bandwidth savings of carrying voice-over-packet networks.

Traditional public switched telephone networks (PSTNs) have long addressed

the voice-quality problem by optimizing their circuits for the dynamic range of

the human voice and the rhythms of human conversation. PSTNs have evolved to

provide an optimal service for time-sensitive voice applications that require

low delay, low jitter and constant, but low, bandwidth. PSTN voice quality is

relatively standard and predictable.

IP networks, however, were built to support non-real time applications, such

as file transfers or e-mail. These applications are characterized by their

bursty traffic and sometimes high bandwidth demand but are not sensitive to

delay or delay variation.

Voice quality can be improved in two ways — special quality of service

algorithms and more bandwidth. The current research on improving QoS in packet

networks is directed at enhancing routing capabilities. Besides, billions are

being spent on adding more bandwidth capacity to global data networks. These

have the potential to make IP telephony a viable commercial alternative to the

PSTN.

Delay

Delay, or latency, is the major challenge to QoS for packet voice networks.

It is the time required for transmission of data packets from origin to

destination. Delay occurring on IP networks primarily results from bandwidth

sharing and processing at routers and endpoints within the networks. Data

applications, for which IP networks were originally designed, are more tolerant

of delay than voice. Data transmission such as e-mail can have an accepted

delivery time of several hours. Even more real time applications such as

database queries can tolerate several seconds of delay.

Delay causes two problems: echo and talker overlap. Echo is caused by the

signal reflections of the speaker’s voice from the far-end telephone equipment

back into the speaker’s ear. Echo becomes a significant problem when the

round-trip delay becomes greater than 50 milliseconds. As echo is perceived as a

significant quality problem, voice-over-packet systems must address the need for

echo control and implement some means of echo cancellation. Talker overlap (or

the problem of one talker stepping on the other talker’s speech) becomes

significant if the one-way delay becomes greater than 250 milliseconds. The

end-to-end delay budget is therefore the major constraint and driving

requirement for reducing delay through a packet network.

Jitter

Jitter, or delay variation, is the result of packets arriving at their

destination at irregular intervals. Bursts of Internet traffic create jitter

problems. This distortion is particularly damaging to the QoS of VoIP

applications. Severe jitter in IP voice transmissions causes jittery or shaky

voice quality.

Removing jitter requires collecting packets and holding them long enough to

allow the slowest packets to arrive in time to be played in the correct

sequence. This causes additional delay.

Packet Loss

Packet network applications compensate for packet loss by retransmitting lost

packets through the use of transmission control protocol (TCP). Data

applications such as file transfers and e-mail are less sensitive to the time it

takes for this to occur, but real-time voice traffic cannot tolerate this delay.

In addition, VoIP networks use connectionless transfer protocols such as user

datagram protocol (UDP) that do not guarantee delivery at all.

Network congestion results in dropped packets. In current IP networks, all

voice frames are treated like data. Under peak loads and congestion, voice

frames will be dropped equally with data frames. Data frames are not time

sensitive and dropped packets can be appropriately corrected through the process

of retransmission. Lost voice packets, however, cannot be dealt with in this

manner. Lost packets mean lost voice information.

Echo Compensation

Echo in a telephone network is caused by signal reflections generated by the

hybrid circuit that converts between a four-wire circuit (a separate transmit

and receive pair) and a two-wire circuit (a single transmit and receive pair).

These reflections of the speaker’s voice are heard in the speaker’s ear.

Echo is present even in a conventional circuit-switched telephone network.

However, it is acceptable because the round-trip delays through the network are

smaller than 50 milliseconds and the echo is masked by the normal side tone

every telephone generates. Echo becomes a problem in VoIP because the round-trip

delay through the network is almost always greater than 50 milliseconds.

QoS in IP networks and the public Internet is expected to improve with

innovations in routing protocols and improvement in physical networks that carry

IP traffic. However, the poor quality of telecommunication infrastructure and

congestion in the IP infrastructure in developing countries would continue to

haunt voice over packet networks for a long time to come.