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E-ISSN: 2348-0831 Volume 1, Issue VI (SEPT-OCT 2014) PP 12-18

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IP PBX: ARCHITECTURE AND PROTOCOLS

Mr. Mukund A. Ghogale1, Dr. Prashant V. Ingole2

Department of Electronics & Telecommunication G.H.Raisoni College of Engineering &

Management1
Principal, G.H.Raisoni College of Engineering & Management, Amravati (MS) 2
mukundghogale@gmail.com1, prashant.ingole@raisoni.net2

Abstract- Wireless communication plays a vital role in day to day life. Voice over IP (VoIP) is one of the most exciting new
developments emerging within the telephony market. Wireless VoIP utilizes wireless LAN technology, the same wireless
infrastructure used for the corporate network, in order to communicate. Traditional PBX systems can be replaced by IP PBX
system in a cost effective way. Wireless IP phones can be used to access corporate telephony system as this technology
combines the telephony function directly into an already existing data network infrastructure. This paper discusses the
working and principle of wireless IP telephony.

Keywords- wireless communication, VoIP, wireless LAN, PBX, IP phone.

1. INTRODUCTION

IP Telephony or VoIP (Voice over IP) allows users to transmit voice and fax or simply “phone calls”
over data networks using the Internet Protocol (IP). IP telephony is sometimes confused with Internet telephony,
but IP telephony is a different term, as it includes Intranet and Extranet communication. This is most interesting
for companies or institutions which have several branches or for mobile employers who are moving inside the
intranet with their mobile devices. On the other hand ,the huge investments that have to be done by the
companies or institutions to purchase a Private Branch Exchange (PBX) can be reduced by using PBXs based on
free software. They provide the same functionality as a traditional PBX.

One of the greatest benefits of the wireless IP phone is that it allows you to carry your office extension
with you inside a wireless networked environment. Unlike your cell phone, the wireless IP phone is part of your
corporate phone system, and carries your personal extension and the same features that your office phone
system has.

Most of the interest in Internet telephony is motivated by cost savings and ease of developing and
integrating new services. Internet telephony integrates a variety of services provided by the current Internet and
the Public Switched Telephone Network (PSTN) infrastructure.

Transmission of voice over internet has certain advantages, such as efficient use of existing networks,
especially for corporations and companies with large volumes of international voice traffic. Home users can also
benefit from making international VoIP calls at much lower rates than any other POTS (Plain Old Telephone
Service) provider can deliver. Another benefit of IP telephony is the relatively easy establishment of new service
prodders, as they can use existing telecommunication networks.

2. OVERVIEW OF VOIP SIGNALLING PROTOCOLS

In this section a brief description of the H.323, SIP and MGCP protocols are given, defining their main
components and functionalities.

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H.323: The H.323 is an ITU-T standard for multimedia communications over local area networks that do not
guarantee Quality of Service (QoS). It addresses call control, multimedia management, bandwidth management
and interfaces between LANs and other network[6]. H.323 defines how voice, video and data traffic should be
transported over IP-based LANs. It also includes the ITU-T Recommendation T.120 data-conferencing
standard. The H.323 recommendation is based on RTP/RTCP (real-time protocol/real-time control protocol) for
managing audio and video signals.

What sets H.323 apart is that it addresses core Internet applications by defining how delay- sensitive
traffic such as voice and video get priority transport to ensure real-time communication service over the Internet.
Related protocols are ITU-T Recommendation H.324 specification, which specifies the transport of voice, data
and video over regular telephone networks. Another related protocol is ITU-T Recommendation H.320, which
defines the transport of voice, video and data over the integrated services digital network (ISDN).

H.323 deals with three basic functional entities of VoIP, these are:

 Media Gateway

 Multipoint Control Unit (MCU)

 Gatekeepers

Media gateway performs the translation of signalling and media exchange between H.323 and PSTN endpoint.
MCUs are in practice part of a gatekeeper or a high speed computer which acts as a terminal that serves one or
more users. Gatekeepers are responsible for performing call authorization, address resolution and bandwidth
management. The H.323 standard defines procedures for user registration with a gatekeeper, call control and
logical channel capabilities negotiation between two or more parties. The Registration,
Authentication/Admission and Status (RAS) and the call control are as per the standard H.225. The logical
channel capabilities negotiation is done as per the H.245 standard. Usually in real-time communication between
two endpoints, the two endpoints are H.323 terminals. But, in general, the communication could be in a
dissimilar domain – PSTN, SIP or MGCP, using a gateway.

SIP : The Session Initiation Protocol [1] is a text-based control (or signaling) protocol similar to HTTP. It is a
client server protocol which can be transported over TCP or UDP, but SIP is commonly implemented over UDP
for simplicity and speed [6]. SIP is more simpler than H.323 protocol. The SIP comprise of user agents and one
or more servers. The entity which initiates SIP request is called a SIP client and the responding entity is called a
SIP server. Initially a signalling connection between SIP client and server is opened. SIP call control uses
Session Description Protocol (SDP) to describe the details of the call (i.e. audio, video, a shared application,
type of codec, size of packet, etc.). SIP makes use of Uniform Resource Identifier (URI) for identification of
logical destination, not an IP address. The address could be a nickname, an e-mail address, or a telephone
number. Other than setting up a phone call SIP can notify users of events, “I am online”, or “e-mail has arrived.”
Instant text messages can also be sent by using SIP.

Using a client-server model, SIP defines logical entities that can be implemented separately or together in the
same product. Client send SIP requests, whereas servers accepts SIP request, execute it and respond. SIP defines
six request methods :

REGISTER allows either the user or a third party to register contact information with SIP server. INVITE
initiates the call signalling sequence. ACK and CANCEL support session setup. BYE terminates a session.
OPTIONS queries server about its capabilities.

Some of the important SIP functional entities are:

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 User Agent can act as both, a SIP client when a SIP request is initiated and a SIP server when SIP
request is received.

 SIP Proxy act as a SIP client and SIP server in making SIP requests in behalf of the other SIP clients.

 Registrar is a SIP server that receives, authenticates and accepts register requests from SIP clients. It
may be allocated with a SIP proxy server.

 Location Server is used to store user information in a database and it helps determine where (to what IP
address) to send a request. It may be allocated with a SIP proxy server.

 Redirect Server is stateless. It just responds to a SIP request with an address.

MGCP : MGCP is master/slave protocol. The call processing functions are separated from gateway functions
by defining a special entity called “call agent”. The call agent is used to control the media gateway. It need not
be located near the gateway. Such setup simplifies the VoIP gateway products, allowing gateways to be located
in home and offices in a cost effective way.

MGCP also uses the Session Description Protocol (SDP) to define the media aspects and other
parameters to initiate establishment of connection between endpoints. MGCP is designed as a distributed system
protocol that give the user the appearance of single VoIP system. It is a stateless protocol in the sense the
sequence of transactions between media gateway and call agent can be performed without track of previous
transactions.

MGCP support following media gateway functions :

 Create, modify and delete connections using any combination of transit network, including frame relay,
ATM, TDM, Ethernet or analog.
 Collect digits as per the digit map received from the call agent, and send complete set of dialled digits to the
call agent.
 Report call statistics.
 Allow mid-call changes, such as call hold, playing announcements and call conferencing.
 Detect or generate events on end points or connections.
Connections are created on the call agent at each end point that will be involved in the call. When the two
endpoints are located on gateways that are managed by the same call agent, the creation is done via the
following three steps:
 The Call Agent asks the first gateway to create a connection on the first endpoint. The response sent by the
gateway includes a session description that contains pertinent information required by third parties to be
able to send packets to the new connection that has been created.
 The Call Agent then sends the session description of the first gateway to the second gateway and asks it to
create a connection on the second endpoint. The second gateway responds by sending its own session
description.
 The Call Agent uses a modify connection command to provide this second session description to the first
endpoint. Now communication can occur in both directions.

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Figure 1 : IP PBX Architecture

3. ARCHITECTURE: IP PBX

Figure 1 shows the architecture of SIP based PBX test bed and interaction among different system
components .

3.1 Components

SIP Server : sipd is a SIP proxy, redirect and reregistration server.

Conference Server : sipconf is a centralized audio/video conference server.

Media Server : rtpd is our general purpose streaming media server, which is used for the storage and delivery
of announcements and VoIP messages.

PSTN Gateway : A router with SIP/PSTN capability is connected to the departmental telephone switch (PBX)
with a T1 trunk and to the department LAN.

Unified messaging: sipum is a centralized answering machine and voice mail system that uses rtspd for storing
announcements and messages.

Protocol converter: Protocol converter is a signalling gateway between different flavours of VoIP protocol.
SIP(Session Initiation Protocol) and H.323 are the two basic VoIP signalling protocols used worldwide. H.323
is ITU-T's standard for multimedia conferencing over any packet based network. SIP is the IETF protocol for
VOIP and other text and multimedia sessions.

4. VOICE PACKETIZATION

Voice is an analog signal. Data protocols like IP transport information as a binary bit stream. Since
binary bit streams are digital signals, before voice can be sent over an IP telephony network, it must first be
converted into a digital format. The two methods of doing this are waveform coding and vocoding[7].
Waveform coding has been used in switched telephony applications since the earliest introduction of digital
multiplexing in the mid-1960s. It is a relatively straightforward technique that converts time-based samples of
speech waveforms into groups of eight binary digits. Vocoding, on the other hand, is a relatively new

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technology, which creates a different type of digital bit stream from models of the frequencies found in speech
Both these process of converting an analog voice signal to a digital bit stream involve the three steps of
sampling, quantizing, and encoding the signal. For IP telephony to work, the voice signal must also be
compressed and packetized. Each of these steps is explained below.

Sampling: A mathematical proof in communications theory called the Nyquist theorem states that the entire
information content of a signal is preserved if the signal is sampled at twice the rate of its highest frequency
component. For voice over a telephone network, filters limit the upper value of frequencies to just under 4,000
Hertz. The voice telephony signal must, therefore, be sampled at 8,000 times per second to preserve its quality.

Quantizing: Quantizing takes the sample value and assigns it to a parameter that most closely describes it. The
way this is done depends on whether waveform coding or vocoding is being used. With waveform coding, the
magnitude (voltage level) of the PCM sample is assigned to the closest integral value of the encoding scale.
Thus, a sample value of 99.5 units may be assigned a quantum value of 100. Quantizing for vocoding is a more
complex process. Samples are grouped into frames of data, and a mathematical process called Discrete Fourier
Transform is used to change the samples from their transform is used to change the samples from their
representation as a signal magnitude at a particular time into a corresponding representation as a set of
frequencies. Parameters describing the frequency representation are derived, and these parameters are then
quantized.

Encoding: Encoding is the process that creates the digital bit stream that represents the signal. In systems that
use waveform encoding, once again, this is a straightforward process that changes the decimal value of the
quantum to its binary equivalent. In systems that use vocoding, encoding includes determining the order in
which the parameters are transmitted in the bit stream. In literature on vocoding systems, encoding is often the
term used to describe everything that happens to the signal after it has been sampled in order to create a digital
bit stream, including any quantization.

Compression: Without compression, the required bandwidth of a digitized voice signal is 8,000 x 8 bits per
second, or 64 Kbps. Circuit switched telephony, using time division multiplexing, has the capacity to process
and transport voice at this rate because of dedicated paths through the public switched telephone network.
Because packet switched telephony is designed to share the physical connections, voice packet calls with this
bandwidth would rapidly consume existing network resources (total bandwidth of the physical connections
between data nodes). Compression increases the number of voice conversations that can be carried by a packet
network at a given data rate.
VoIP Codecs Compressed Voice Digitizing Complexity
Rate (Kbps)
G.711 PCM 64 NA
G.726 ADPCM 40/32/24 Low (8 MIPS)
G.729 CS-ACELP 8 High (30 MIPS)
G.729A CA-ACELP 8 Moderate
G.723 MP-MLQ 6.4/5.3 Moderate-High (20 MIPS)
G.723.1 MP-MLQ 6.4/5.3
G.728 LD-CELP 16 Very High (40 MIPS)
Table 1 : Compression Standards

Packetizing : Packetizing is the grouping of the bits within the binary bit stream into data units consisting of a
payload section and a header. Packets operate at the third layer of the OSI protocol reference model to route
information across a data network. The header contains useful overhead information, such as originating and
terminating addresses, routing priorities, and error detection mechanisms. The payload section contains the
encoded voice information.

01100010101001

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Header Payload
Figure 2 : Packetization

Packet size is one factor that affects the number of calls that an IP network can carry. For a typical CODEC
using G.729 compression, the voice information in a 35-millisecond “talk spurt” requires 40 bytes. This is only
the payload part of the packet.

Only 8 Kbps of bandwidth is required to transport these 40 bytes.(40 bytes are 40 samples in a T1 signal).
There are a total of 8,000 samples per second in T1. 40/8,000 x 1,540,000 bits per
second = 7.7 Kbps.) Headers must be added to the voice information for control and routing. The corresponding
length of header for the 40 byte, 35 msec. “talk spurt” in our example is 28 bytes for the IP/UDP protocol
overhead, and 6 bytes for the voice payload overhead, for a total header overhead of 34 bytes. This overhead
adds about 7 Kbps to the 8 Kbps voice payload, resulting in a total bandwidth of 15 Kbps.

28 Bytes 6 Bytes 40 Bytes

IP/UDP Voice Voice

Header Header Information
Figure 3 : IP/UDP Voice Packet

Compression is not limited to packet switched networks. The telephone industry has used a type of
compression known as Adaptive Differential Pulse Code Modulation (ADPCM) in toll networks within the
Public Switched Telephone Network for a number of years. ADPCM, also known as ITU G.724, codes only the
difference in samples, and uses a four bit-coding scheme. The result is a 32 Kbps bandwidth requirement per
voice call—still high for IP telephony.

5. CONCLUSION

In this paper, a simplex architecture of IP PBX system is being discussed. Different protocols
interconnection scenarios as well as variety of voice codecs also been discussed. It is a real challenge to provide
a reliable and high-quality network for data communication between different end points. Factors involved in
designing a high-quality IP PBX system include the choice of codec and call signalling protocol.

We have compared several VoIP call signalling protocols. A peer-to-peer control signalling paradigm
is used by SIP and H.323, while MGCP uses a master-slave signalling paradigm. In early phase H.323 had the
lead among VoIP services, but SIP is becoming more popular. MGCP is appropriate for the control of many
low-cost IP telephony residential gateways. For communication among call agents, SIP may be more
appropriate. SIP also offers a wide range of services beyond basic telephony including call parking, presence
management and IVR services.

Finally we have discussed how the speech signal is converted into the packets by the process of
packetization. Compression procedures and techniques are also given. Variety of VoIP codecs has been
compared on the basis of data rates and complexity of implementation.

There is still some development to be done in the subject. There is still much confusion on what should
be the best standard and what protocols should be used. Interconnection of protocols is not always as easy as it
appears to be. Many of them still lack some features or have certain disadvantages that are difficult to be
conquered.

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6. REFERENCES

[1] Bur Goode, Senior member, IEEE, ”Voice Over Internet Protocol(VoIP)”, in Proceedings of the IEEE, Vol.
90, No. 9, September 2002

[2]Alec Vugrinec, Student Member IEEE, Saso Tomazic, Member IEEE,” IP telephony from a user
perspective”, in 10th Mediterranean Electro technical Conference, MEleCon 2000, Vol. I

[3]Miguel Edo, Miguel Garcia, Carlos Turro and Jaime Lloret,” IP Telephony development and performance
over IEEE 802.11g WLAN”,in Fifth International Conference on Networking and Services, 2009

[4]Mohsen Gerami,”Wireless IP telephony”, in (IJCSIS) International Journal of Computer Science and

Information Security, Vol. 7, No. 2, 2010

[5]M. Bearden, L. Denby, B. Karac¸alı, J. Meloche, D. T. Stott, “Assessing Network Readiness for IP
Telephony”.

[6]Anoop Kumar K. and Tanu Malhotra, Wireline Systems Group, R&D Center, ”A Multi-Signalling Protocol
Architecture for Voice over IP Terminal”, in IEEE Infocom, 2004

[7] DigiPoints, the digital knowledge handbook, Volume III,Issue 11, November 2006.

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