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For years, many different data networking
protocols have existed, but now, data communications has firmly
found its home in the form of IP, the Internet Protocol. On the
even since telecommunications began, voice communications has
rigidly stuck to a classic circuit switched networking environment.
Over the years, many technologies have been used for the integration
of voice and data with some offering much better solutions than
others, at least from a technical perspective. IP is far from
the best technical solution for the support of voice, however
there are many reasons why it is rapidly becoming the
platform of choice for, not only data, but voice too.
This course looks at the subject of voice over IP, covering
various methods of its implementation and looking at many of
the relevant standards. We also cover many of the issues that
need to be considered when deploying voice over IP and discuss
how these may be dealt with.
In order to present the information in such a way that
it is best understood, we make significant use of live demonstration
equipment to provide a real-life and tangible perspective
to the course. This course covers live demos of IP phones,
IP PBX systems, gateways, voice compression systems and covers
both the H.323 and SIP protocols. We also make use of protocol
analysis equipment to show what is actually
the IP network.
Who should attend the course?
This course is aimed at voice and/or data professionals who
wish to extend their knowledge and experience into the area of
voice over IP. A detailed knowledge of neither voice nor data
(IP) is required for a student to derive benefit from the course.
A general understanding of voice and/or data (IP) will however
be of help.
Introduction to voice over IP (VoIP)
This section gives a basic introduction to voice over IP, looking
at some fundamental applications and business reasons as to why
one might consider deploying it.
- Why VoIP? A view from a business perspective as to why voice
over IP may be an appropriate technology to deploy in many
voice related applications
- A look at voice over IP as deployed across the Internet or
across a private IP network.
- An introduction to some of the standards for voice over IP:
- H.323 and related protocols
- Session Initiation Protocol (SIP)
- Media Gateway Control Protocol (MGCP) and Megaco / H.248
- An introduction to some of the devices that may be used to
implement a voice over IP network
- IP phones
- Power to the IP phone - powered Ethernet
- Call control systems (e.g. the Gatekeeper)
- IP enabled voice switches (PBX and public network switches)
- An introduction to some of the challenges of deploying voice
Overview of the voice technologies relevant to voice over IP
This section gives a simple overview of the voice related technologies
that we see as relevant to voice over IP. It is not our intention
to provide an in-depth voice technology course coverage as this
would take too long. Having said that, the course tutor is very
experienced in voice technology training and, if appropriate,
can spend extra time with students if required to help them understand
the important aspects (This typically would be done during a
break so as not to impact others on the course).
- Interface types:
- Foreign Exchange Station (FXS) and Foreign Exchange Office
- Ear & Mouth - E&M 2 wire and 4 wire
- 2.048Mbit/s digital interface - G.732/G.704/G.703
- Channel Associated Signalling ( CAS).
- Common Channel Signalling (CCS) with a summary of some
of the key types including Q.931, QSIG, DPNSS, SS7
- Voice Coding:
- G.711 Pulse Code Modulation (PCM) - 64kbit/s
- Compressed voice:
- G.728 Low-Delay Code Excited Linear Prediction (CELP)
- G.729 Conjugate Structured Algebraic CELP (CS-ACELP)
- Also G.729a and b
- G.723.1 MP-MLQ and ACELP
- The use of Mean Opinion Score (MOS) to quantify the quality
of voice communications
Overview of the IP technologies relevant to voice over IP
This section gives a simple overview of the IP related protocols
that are relevant to voice over IP. It is not our intention to
provide an in-depth IP course coverage as this would take too
long. Having said that, the course tutor is very experienced
in IP technology training and, if appropriate, can spend extra
time with students if required to help them understand the important
aspects (This typically would be done during a break so as not
to impact others on the course).
- The IP protocol and the key parts of the IP header relevant
to voice over IP. We discuss this in relation to IP version
4 and IP version 6
- Overview of IP addressing and network and subnet addressing
- Other key protocols:
- Transmission Control Protocol (TCP)
- User DataGram Protocol (UDP)
- Real-Time Protocol (RTP) and Real-Time Control Protocol
The H.323 framework for voice over IP
H.323 is the oldest and most commonly used protocol suite used
to deliver voice over IP. The reasons for this are manifold including
fact that the protocol specifications have been in place for
a long time and others such as Microsoft's inclusion of H.323
in its Netmeeting software, an application residing on virtually
every desktop computer in the world. The original version (V1)
of H.323 released in 1996 was referred to as "Visual telephone
systems and equipment for local area networks which provide a
non-guaranteed quality of service". Its scope goes far beyond
just voice over IP and in recognition of its expanded scope to
cover not just LANs, it was renamed in version 2 as "Packet
based multimedia communications systems". Since then, it has
evolved further, adding various new features and improvements plus
correcting issues that were present in its earlier versions.
- H.323 architecture:
- The H.323 Terminal
- The H.323 Gateway
- The H.323 Gatekeeper
- The H.323 Multipoint Control Unit (MCU)
- H.225.0 - This deals with a number of aspects of establishing
communication across the data network including Registration,
Admissions & Status (RAS) operations, along with Q.931
signalling messages for call setup, call control, and communications
between terminals, gateways, gatekeepers, and MCUs.
- H.245 - This specifies the syntax and semantics
of terminal information messages as well as procedures to use
them for in-band negotiation at the start of or during communication.
Session Initiation Protocol - SIP for voice over IP
The Session Initiation Protocol (SIP) is a signalling protocol
used for establishing sessions in an IP network. A session could
be a simple two-way telephone call or it could be a collaborative
multi-media conference session. It is basically a very simple
developed purely as a mechanism to establish sessions. It does
not know about the details of a session, it just initiates, terminates
and modifies sessions. In order to handle other aspects of end-to-end
communications, SIP was designed to
reuse many existing
- SIP architecture:
- SIP user agents (user devices) - phones, workstations,
- SIP servers - Proxy server, redirect server, registrar
- SIP location servers
- SIP gateways
- SIP functions:
- Address resolution
- Session setup
- Media negotiation using the Session Description Protocol
- SIP examples - We shall look at a number of examples of SIP's
use including basic call setup, call forward plus others.
MGCP and Megaco / H.248 for voice over IP
Media Gateway Control Protocol (MGCP) and Megaco / H.248 are
control protocols designed to be used between a Media Gateway
Controller or call agent and a media gateway. Megaco / H.248
is newer than MGCP, and is seen by many as MGCP's replacement.
That said, many products already employ MGCP and therefore, for
some time, both protocols will be used.
- Media Gateway Controllers
- MGCP & Megaco / H.248 message examples
Voice over IP applications
Voice over IP may be applied in so many ways that it makes it
very difficult to cover all possible applications. This section
aims at showing how voice over IP is, and will be in the future,
used in various real-life applications. It also shows some of
the additional benefits that integrating voice into an IP environment
may offer over and above the classic way of supporting voice
with a standard voice switch.
- PBX replacement applications - Full-blown IP telephony using
IP phones, call control platforms, gateways etc.
- Toll bypass - Using voice over IP gateways to allow long-distance
voice communications across existing data networks including
private data networks and the Internet, thus bypassing the
call charges normally associated with routing calls via the
public telephone network.
- IP-enabled PBXs - Typically seen as an interim option between
a traditional PBX based system and a full-blown IP solution.
This option typically takes a PBX and adds IP capabilities
so that PBX users can communicate with other IP telephony users.
QoS requirements and solutions with voice over IP
Often banded about as the reason why voice over IP will not
work successfully, quality of service is key to the operation
of a real quality-grade voice over IP network. Many technologies
and protocols exist to solve the quality of service issue, and
here we intend to explain the key ones and de-mystify any uncertainties
about how voice over IP may be deployed and operate very successfully.
- Quality of Service (QoS) issues:
- Basic voice quality in terms of distortion, frequency
- The causes of delay in a voice over IP system
- A description and explanation of what levels of delay
will actually impair a voice conversation and what levels
will not be noticed
- The effect of delay resulting in echo
- The effect of transporting non-voice signals such as
fax, modems, DTMF tones etc.
- Description of some protocols used to help deliver QoS guarantees:
- IEEE 802.1p / IEEE 802.1Q
- Asynchronous Transfer Mode (ATM)
- Resource Reservation Protocol (RSVP)
- Differentiated Services (Diffserv)
- Multi-Protocol Label Switching support of QoS
considerations with voice over IP
We finish up with a look at a few potential issues with using
voice over IP.
- Network Address Translation (NAT) issues
- Issues with voice over IP across Firewalls