3. Table of Contents
Overview of VoIP………………………………………………………………………...4
VoIP Components:
Terminals ............................................................................................................................5
Gateways.............................................................................................................................6
Gatekeepers.........................................................................................................................7
Multipoint control unit……………………………………………………………………7
VoIP Protocols:
H-323 .................................................................................................................................8
Session initiation protocol……………………………………………………………......10
VoIP Signaling and routing …………………………………………………………....12
Benefits and requirements of VoIP….………………………………………………....15
Conclusion………………………………………………………………………………17
References………………………………………………………………………………18
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4. VOICE OVER INTERNET PROTOCOL
OVERVIEW OF VoIP
Voice over IP is the transport of voice using the Internet Protocol (IP) however this broad term
hides a multitude of deployments and functionality. So voice over Internet Protocol is a method
for taking analog audio signals and turning them into digital data that can be transmitted over the
Internet.
The following types of VoIP applications are in use:
ATA
The simplest and most common way is through the use of a device called an ATA (analog
telephone adaptor). The ATA allows you to connect a standard phone to your computer or your
Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the
analog signal from your traditional phone and converts it into digital data for transmission over
the Internet.
IP Phones
These specialized phones look just like normal phones with a handset, cradle and buttons. But
instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet
connector. IP phones connect directly to your router and have all the hardware and software
necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make
VoIP calls from any Wi-Fi hot spot.
Computer-to-computer
This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls.
There are several companies offering free or very low-cost software that you can use for this type
of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet
connection.
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5. VoIP COMPONENTS
TERMINALS
IP Phones
An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP
network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet,
or a private IP Network such as that of a company. The phones use control protocols such as
Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary
protocols such as that used by Skype. IP phones can be simple software-based Soft phones or
purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone.
Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA). H.323
protocol and provide real-time, two-way multimedia communications. In the case of voice, the
H.323 terminal is generally an IP telephone.
Analog Phones
A telephone can be a basic push-button wall unit or an integrated system complete with
answering machine, stored-number dial, speaker phone, and 900MHz cordless operation.
Computers
With VoIP software such as Skype, yahoo, netmeeting and many more running on computers
they can also be used as communication devices. H.323 is also widely deployed on PCs. A very
common application of the H.323 protocol can be found in the Microsoft NetMeeting software
that allows for both voice and video transmissions on a user’s PC.
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6. Gateways
Gateways work as a translator to allow communications between H.323 and non-H.323 entities
(for instance, between H.323 terminals and telephones on the circuit-switched network). H.323
gateways provide a means for an H.323 network to communicate to other networks, most
typically the PSTN or PBX systems. In order to provide this interoperability, gateways provide
for translation and call control functions between the two dissimilar network types. Encoding,
protocol, and call control mappings occur in gateways between two endpoints. Gateways provide
many functions, including:
Translating protocols
The gateway acts as an “interpreter,” allowing the PSTN and the H.323 network to talk to
each other to set up and tear down calls.
Signaling Gateway
The Signaling Gateway is located in the service provider’s network and acts as a gateway
between the call agent signaling and the SS7-based PSTN. It can also be used as a signaling
gateway between different packet based carrier domains. It may provide signaling
translation, for example between SIP and SS7 or simply signaling transport conversion e.g.
SS7 over IP to SS7 over TDM.
Trunking Gateway
The Trunking Gateway is located in the service provider’s network and as a gateway between
the carrier IP network and the TDM (Time Division Multiplexing)-based PSTN. It provides
transcoding from the packet based voice, VoIP onto a TDM network. Typically, it is under
the control of the Call Agent / Media Gateway Controller through a device control protocol
such as H.248 (Megaco) or MGCP.
6
7. Access Gateway
The Access Gateway is located in the service provider’s network. It provides support for
POTS phones and typically, it is under the control of the Call Agent / Media Gateway
Controller through a device control protocol such as H.248 (Megaco) or MGCP.
Subscriber Gateway
The Subscriber Gateway is located at the customer premises and terminates the WAN (Wide
Area Network) link (DSL, T1, fixed wireless, cable etc) at the customer premises and
typically provides both voice ports and data connectivity. Usually, it uses a device control
protocol, such as H.248 (Megaco) or MGCP/NCS, under the control of the Call Agent. It
provides similar function to the Access Gateway but typically supports many fewer voice
ports.
Gatekeepers
Gatekeepers provide call control functions such as address translation and bandwidth
management and are often considered to be the most important component in the H.323 stack.
Gatekeepers in H.323 networks are optional. However, if they are present, it is mandatory that
endpoints use their services. The H.323 standards define several mandatory services that the
gatekeeper must provide and specify other optional functionality.
Multipoint Control Units
MCUs provide conference facilities for users who want to conference three or more endpoints
together. MCUs provide a unique function to the H.323 protocol in that they do not provide a
direct interconnection to the H.323 protocol stack. Rather, they provide a method for H.323 to
interconnect voice and videoconferencing. MCUs provide conference support for three or more
endpoints. All terminals participating in the conference establish a connection with the MCU. It
manages conference resources and negotiations between endpoints to determine which audio or
video codec to use.
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8. VoIP PROTOCOLS
H.323
H.323 is probably the most important standard supporting packetized voice technology. H.323 is
an ITU-T recommendation umbrella set of standards that defines the components, protocols, and
procedures necessary to provide multimedia (audio, video, and data) communications over IP-
based networks. Essentially, H.323 provides a method to enable other H.32X-compliant products
to communicate. In addition to control and call setup standards, H.323 encompasses protocols for
audio, video, and data as follows:
Audio
The compression algorithms H.323 supports for audio are all proven International
Telecommunications Union (ITU) standards (G.711, G.723, and G.729). Because audio is
the minimum service provided by the H.323 standard, all H.323 terminals must have
support for at least one audio codec support, as specified by G.711.
Video
Video capabilities for H.323 are optional. However, any video enabled H.323 terminal
must support the ITU-T H.261 encoding and decoding recommendation.
Data
H.323 references the T.120 specifications for data conferencing. An ITU standard, T.120
addresses point-to-point and multipoint data conferences. It provides interoperability at the
application, network, and transport levels.
The H.323 Protocol Stack
Just as with the TCP/IP protocol, the H.323 protocol is actually a suite of protocols that work
together to provide end-to-end call functionality in a converged network. However, the H.323
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9. protocol also relies heavily on the services provided by other protocols such as TCP, IP, and
UDP as well as RTP. The protocols that make up the H.323 protocol are Registration,
Admission, and Status (RAS), H.245, and H.225.
Codecs
Coder/decoders (codecs) are used by not only the H.323 protocol but by all VoIP protocols to
define the degree of compression and decompression algorithms that will be used to transport
either a voice or video transmission across a converged network.
Speech codecs, sometimes called voice encoders or vocoders if source codecs are
used, can be divided into three basic classes: waveform, source, and hybrid.
Waveform codec
These are older, operationally used high bit rates and provide very good quality
speech reproduction.
Source codec
These operate at very low bit rates but tend to produce speech that sounds artificial or
tinny.
Hybrid codec
These use techniques from both source and waveform coding, operate at intermediate
bit rates, and provide good-quality speech.
Fig No 8.1 - The H.323 Protocol Stack
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10. Session Initiation Protocol
SIP is a simple signaling protocol used for Internet conferencing and telephony. SIP is fully
defined in RFC 2543. Based on the Simple Mail Transport Protocol (SMTP) and the Hypertext
Transfer Protocol (HTTP), SIP was developed within the IETF Multiparty Multimedia Session
Control (MMUSIC) working group. SIP specifies procedures for telephony and multimedia
conferencing over the Internet. SIP is an application-layer protocol independent of the
underlying packet protocol (TCP, UDP, ATM, X.25). SIP is based on a client/server architecture
in which the client initiates the calls and the servers answer the calls. Because it is an open
standard based protocol, SIP is widely supported and is not dependent on a single vendor’s
equipment or implementation. However because of its simplicity, scalability, modularity, and
ease with which it integrates with other applications, this protocol is attractive for use in
packetized voice architectures.
Some of the key features that SIP offers are:
Address resolution, name mapping, and call redirection
Dynamic discovery of endpoint media capabilities by use of the Session Description
Protocol (SDP)
Dynamic discovery of endpoint availability
Session origination and management between host and endpoints
SIP has learned from HTTP and SMTP and has built a rich set of extensibility and
compatibility functions.
SIP was designed to be highly modular. A key feature is its independent use of protocols.
SIP has the capability to integrate with the Web, e-mail, streaming media applications,
and other protocols.
.syngress.com
Session Initiation Protocol Components
The SIP system contains two components:
User agents
Network servers
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11. A user agent (UA) is SIP’s endpoint, which makes and receives SIP calls. The client is called the
user agent client (UAC) and is used to initiate SIP requests.
The server is called the user agent server (UAS), receiving the requests from the UAC and
returning responses for the user. There are three kinds of SIP servers:
Proxy server
Proxy servers decide to which server the request should be forwarded and then forward
the request. The request can actually traverse many SIP servers before reaching its
destination. The response then traverses in the reverse order. A proxy server can act as
both a client and server and can issue requests and responses.
Redirect server
Unlike the proxy server, the redirect server does not forward requests to other servers.
Instead, it notifies the calling party of the actual location of destination.
Registrar server
Provides registration services for UACs for their current locations. Registrar servers are
often placed with proxy and redirect servers.
Fig No 8.2 - SIP Components
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12. VoIP SIGNALING AND ROUTING
In telephony, the signaling information is used to exchange information between endpoints on a
network to set up, control, and end calls. The signaling method that's used depends on the type of
device that's being used and the type of signaling method that's used by the telephone company.
On the PSTN local loop
An open circuit with no current flowing indicates an on-hook condition (telephone
handset placed in the cradle).
Offhook (telephone receiver off the cradle) is indicated by a closed circuit with current
continuously flowing.
DP and DTMF are the address-signaling methods implemented from telephone to switch
in the telephone network.
Earth and magnet (E&M) signaling is the most commonly utilized method of analog
trunking.
VoIP Signaling
In connectionless network architectures such as IP networks, the responsibility for session
establishment and signaling resides in the end stations. To successfully emulate voice services
across an IP network, enhancements to the signaling stacks are required. Some are:
H.323 agent is added to the router for standards-based support of the audio and signaling
streams.
The Q.931 protocol is used for call establishment and tear-down between H.323 agents or
end stations.
Real-Time Control Protocol (RTCP) provides for reliable information transfer once the
audio stream has been established. A reliable session-oriented protocol such as TCP is
deployed between end stations to carry the signaling channels.
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13. RTP, which is built on top of UDP, is used to transport the real-time audio stream. RTP
uses UDP as a transport mechanism because it has lower delay than TCP and because
actual voice traffic, unlike data traffic or signaling, tolerates low levels of loss and cannot
effectively exploit retransmission.
H.245 control signaling is used to negotiate channel usage and capabilities. H.245
provides for capabilities exchange between endpoints so that codecs and other parameters
related to the call are agreed upon between the endpoints. It is within H.245 that the audio
channel is negotiated.
wyngress.co
VoIP signaling is most commonly used in three distinct areas:
signaling from the PBX to the router
signaling between routers
signaling from the router to the PBX.
Signaling Between Routers and PBXs
When signaling from PBX to router, the user picks up the handset, signaling an off-hook
condition. The connection between the PBX and router appears as a trunk line to the PBX, which
signals the router to seize the trunk. Once a trunk is seized, the PBX forwards the dialed digits to
the router in the same manner the digits would be forwarded to a telephone company switch or
another PBX. The signaling interface from the PBX to the router may be any of the common
signaling methods used to seize a trunk line, such as FXS, FXO, E&M, or T1/E1 signaling. The
PBX then forwards the dialed digits to the router in the same manner as the digits would be
forwarded to a telco switch. The PBX seizes a trunk line to the router and forwards the dialed
digits. Within the router, the dial plan mapper maps the dialed digits to an IP address and
initiates a Q.931 call establishment request to the remote peer router that is indicated by an IP
address.
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14. Fig No 9.1- PBX-to-Router Signaling
Fig No 9.2 - Router-to-Router Signaling
When the remote router receives the Q.931 call request, it signals a line seizure to the PBX. After
the PBX acknowledges this seizure, the router forwards the dialed digits to the PBX and signals
a call acknowledgment to the originating router.
Fig No 9.3 - Router-to-PBX signaling
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15. BENEFITS AND REQUIREMENTS FOR VoIP
For service providers examining the business case for VoIP, the ubiquity of IP as a networking
technology at the customer premises opens the possibility of deploying a vast range of innovative
converged voice and data services that simply cannot be cost effectively supported over today’s
PSTN infrastructure.
IP-based internet applications, such as email and unified messaging, may be seamlessly
integrated with voice application
IP centrex services allow network operators to provide companies with cost effective
replacements for their ageing PBX infrastructure
VoIP services can be expanded to support multimedia applications, opening up the
possibility of cost effective video conferencing, video streaming, gaming or other multi-
media applications.
The flexibility of next generation platforms allows for the rapid development of new
services and development cycles are typically shorter than for ATM or TDM-based
equipment.
VoIP products based on the MSF architecture, unlike legacy TDM switches, often
support open service creation environments that allow third party developers to invent
and deliver differentiated services.
VoIP leverages data network capacity removing the requirement to operate separate voice
and data networks.
IP equipment is typically faster and cheaper than ATM or TDM-based equipment – a gap
that is increasing rapidly every few months.
Re-routing of IP networks (e.g. with MPLS) is much cheaper than, say, SDH protection
switching.
Whatever the justifications, most service providers recognize that VoIP is the direction of the
future – however when looking at a future PSTN scale solution service providers must ensure
that the following key requirements are met to provide equivalence with the PSTN:
Security
Quality of Service
15
16. Reliability
Migration path
OSS support
Billing
Network Interconnection
These issues are by no means simple and in many cases have delayed roll out of VoIP services.
This white paper will look in more detail at these problems and consider at a high level how they
might be addressed.
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17. CONCLUSION
Voice over IP is quickly becoming readily available across much of the world, however many
problems still remain. For the time being transmission networks involve too much latency or
drop too many packets, this effects quality of service sometimes severely deteriorating the
quality of the call. Also VOIP contains many security risks, sending out packets that any person
may intercept. Although VOIP may offer cheaper solutions for many the PSTN offers a high
QoS and greater security that makes up for its higher prices. It is my belief that the telephone
market will continue to be dominated by the PSTN until quality of service and security issues
can be addressed.
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18. REFRENCES
1. www.wikipedia.org
2. www.britanica.com
3. www.computer.howstuffworks.com
4. Data Communications and Networking by Behrouz A. Forouzan
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