Comparison of Voice over IP with circuit switching techniques

Comparison of Voice over IP with circuit switching

techniques

Author
Richard Sinden

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Abstract
Voice-over-IP is a growing technology. Companies are beginning to consider
commercial investments in these services. Will it be able to provide the quality of
service necessary to satisfy customers? If not what will need to be done to remedy the
situation.

Introduction

Circuit Switched Networks, also known as Public Switched Telephone Network

(PSTN) are the major telecommunications networks in the world today. Almost all
phones are connected to a PSTN. The question is, is this all set to change with the
introduction of Voice over IP, which promises lower costs for service provider, and
hence the customer, by using the existing internet infrastructure to provide cheaper
calls. But, is the internet ready to handle this relatively new type of real-time data
traffic.

Overview of Voice over IP

The idea behind Voice over IP(VoIP) is to allow audio and video communications
across IP-based networks, which, of course, includes the Internet. It also allows calls
to be made between an IP-based network, and a Switched Circuit Network(SCN),
such as a PSTN, or ISDN. The system can deal with the translations required between
the 2 networks. It is also able to handle multipoint connections as well as point-point
ones.

The Protocols
There are currently no standard protocol top level protocols for VoIP. There are,
however, two competing protocols which are being developed by different
organisation.
H.323 is being developed by the ITU. This is a multimedia conferencing protocol,
which includes voice, video and data conferencing.
SIP (Session Initiation Protocol) is being developed by the IETF, and is used for the
session initiation rather than dealing with the actual call after it is connected.

H.323

H.323 is a complex protocol, and calls on many other protocols for the handling of
audio and video compression, call control, bandwidth management etc.

Elements of an H.323 system

Terminals – The end points of the network, e.g. Telephones, videophones or PCs.
Gateways – These act as an interface between the IP-based network, and other
networks e.g. PSTN

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Gatekeeper – This is an optional component which can be used to provide call control
services and address resolution for other elements in its ‘zone’.
Multipoint Controllers(MC) – These are used to manage connections between two or
more terminals. These are usually combined into one of the terminals and known as
an Multipoint control unit (MCU).
Call Terminal A/V Gatekeeper
Signalling Control Data Audio Video Control Control
H.225.0 H.245 T.120 G.7xx H.26x RTCP RAS

RTP

IP Multicast

TCP UDP
IP

Figure 1: The H.323 protocol stack

H.225.0 Call control messages, e.g. signalling and registration

H.245 Terminal control, opening and closing of channels, etc.
T.120 Data conferencing
G.7xx Audio codecs at various rates
G.711 PCM, 64kbps uncompressed
G.723.1 MP-MLQ 6.4kbps compression
G.723.1 AC-ELP 5.3kbps compression
G.726 ADPCM 32kbps compression
G.728 LD-CELP 16kbps compression
G.729A CS-CELP 8kbps compression
H.26x Video codecs

Session Initiation Protocol (SIP)

SIP is a far simpler protocol, which has been designed with the idea of providing a
simple way of setting up a call to another person. It provides functions such as
resolving a called party’s address, negotiation of terminal capabilities, and passing
additional information (e.g. CLI).
This protocol has the advantage that the messages are in text format, which although
means more data is being sent, it does mean that the messages are clear and easy to
understand and debug.

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Server 1 3

Server 2
4 6
5 1

2
Terminal 2 7

8 Terminal 1

Figure 2: Overview of a basic SIP system

In this case, to begin with the user at terminal 2 would register themselves with server
1(1). When the user at terminal 1 wanted to contact terminal 2, they would send a
request to server 2. The request would go to server 2(2) which would look in its
database and find that the address for terminal 2 was at server 1, so it would forward
the call to server 1(3), which would again look in its database, and find that Terminal
2 was registered to a specific address, which it would send a call request to Terminal
2(4). When the phone is answered the acceptance message goes all the way back to
Terminal 1(5,6, and 7) where Terminal 1 is then able to directly contact Terminal 2(8)
without the need for the servers any longer. At this point the servers can destroy all
call state information if they wish.
This has the advantage that servers do not need to maintain call state throughout the
call which obviously reduces load.

Potential problems with Voice over IP

There are two main areas where Quality of Service(QoS) needs to be addressed,
the setting up of the call, and the call itself .

Setting up the call needs to be as speedy as possibly, and this relies on being able to
locate the user quickly and efficiently, and keeping any handshaking between
terminals and gateways to a minimum. Generally, a setup time of a few seconds is the
maximum that would be acceptable. Here SIP has the advantage over H.323 despite
its lengthy text based headers, the simplicity of its approach means that there is a
minimum of handshaking and messages passed between servers before a connection
can be established.

The quality of the call itself can be affected by six main factors: latency, bandwidth,
jitter, packet loss, network/service availability and transcoding loss.

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Latency is the delay between the data leaving one terminal and arriving at another.
This affects both VoIP and PSTN. It is agreed that an average person will not notice
any delay with a 300ms round trip time(RTT) for the data. The current
recommendation from the European telecoms industry association, ETSI is that the
round trip time for VoIP should not exceed 200ms.

Bandwidth is probably the major factor in determining the latency of a connection,

provided that efficient routing is in place. Low bandwidth can cause packet queuing
which will obviously increase the RTT.

Jitter is another problem which can be caused by low bandwidth, this is where the
variation in packet delays because too great. Which can cause packets to arrive out of
sequence. Buffering helps to deal with this, but obviously with too much buffering
then the latency will increase.

Packet loss is also caused when the network is congested due to insufficient
bandwidth. Audio and video can handle this to a certain extent, but too much and
quality will be reduced.

Availability of the route after it has been setup is also important. If a route was to
become unavailable during a call then it would be necessary to find a new one, before
too much data was lost.

Transcoding loss is not a problem in a purely IP-based call, but it can be a problem
when switching between analogue and digital networks. This is where to much
encoding and decoding of a audio stream can cause a deterioration in the quality of
the audio.

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Audio Codecs Comparison

A survey by the telecommunications industry association(TIA) returned the following

data on user satisfaction with latency problems and packet loss. A circuit switched
network, obviously does not have to deal with packet loss, but was tested with varying
latencies, and produced the following results.

Figure 3: PSTN performance with increasing latency [TIA 2001]

These results indicate that it is indeed the case that when the RTT begins to rise over
300ms it does become noticeable to the user, as satisfaction levels begin to drop off
quite steeply at this point.

Using VoIP with the G.711 audio codec, as you would expect, without packet loss
then the results are identical to those of the PSTN. Once packet loss is introduced
however, the QoS begins to drop dramatically, especially with no packet loss
concealment in place. Although with the packet loss concealment and packet loss
under 3% and the one-way delay under 100ms as ETSI recommends, the quality is
still in the satisfactory boundary defined by the PSTN (see figure 4).

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Figure 4: G.711 performance with increasing packet loss and increasing latency [TIA 2001]

However, compressed data is another story. As you can see from the diagram with the
G.723.1 at 6.3kbit/s the audio quality is never even reaches the satisfactory level at
0% packet loss.

Figure 5: G.723.1 performance with increasing packet loss and increasing latency [TIA 2001]

The G.729A protocol, which is compressed to 8kbit/s, is slightly better with the user
response residing in the satisfactory zone provided that the RTT is under 200ms and
there is very little packet loss. This still does not provide the same kind of quality as a
circuit switched network though.

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Figure 6: G.729A performance with increasing packet loss and increasing latency [TIA 2001]

Summary

It is clear from these figures that circuit switched networks have a definite advantage
over packet switched networks, by not being affected by packet loss. This is currently
a stumbling block for Voice over IP. Packet loss on the networks needs to be reduced,
and this can be done by reducing congestion. To do this, either bandwidth capacities
need to be increased, or a priority system needs to be developed which allows packets
to be identified as high priority real time traffic. A Type of Service(ToS) field is being
introduced into the IP header in an effort to allow prioritising of packets, but this is
not yet widely supported. In the future maybe this will become a viable technology,
but with the current state of the internet latency and packet loss issues will degrade
quality too much to make it a widely used service.

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References
Telecommunications Industry Association (TIA)(2001) TIA/EIA/TSB116
http://global.ihs.com/search_res.cfm?RID=TIA&INPUT_DOC_NUMBER=TSB%2D116&pa
rtial_match=on&nbr_rows=25

David Willis (1999) Voice Over IP, The Way It Should Be

http://www.networkcomputing.com/1001/1001ws1.html

Camarillo (2001) Signalling in the circuit switched network

http://www.books.mcgraw-hill.com/fatbrain/telecom/features/sept01/camarillo/0-07-
137340-3_Ch01.pdf

Rosenberg et al. (2001) SIP: Session Initiation Protocol

http://www.cs.columbia.edu/sip/drafts/draft-ietf-sip-rfc2543bis-05.pdf

Communicate (2000) Clarity is the best policy

Shara Evans (2000) H.323 Updates CommsWorld

Richard Chirgwin (2000) Real-time Enough? CommsWorld

Paul E. Jones (2001) Current Status of H.323

http://www.h323forum.org/papers/current_status_h323.zip

Boaz Michaely (2000) H.323 Overview

http://people.itu.int/~michaely/docs/indepth_H323.ppt

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