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VoIP Training, Chapter 3 - How Does VoIP Work?

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LRG Networks.com
LWC Training Corp.

Voice over IP Online Course


Lesson 3 - How Does VoIP Work?

Menu B. Call Manager


Home
VoIP Home
Preface
Lesson 3 Intro Call Manager

Lesson Index  VoIP needs a "call manager" to facilitate calls


A. Call control  Direct node to node IP calls are rare
C. Transmitting voice  Must map telephone numbers or names to IP addresses
Review/Exercise  Standalone dedicated box or built into routers
 Uses either standard protocols H.323 or SIP, or proprietary, such as Cisco's
SKINNY

The call manager on an IP Telephone system provides the management of the


telephone calls, similar to the function that SS7 does for the PSTN. Although in
theory two IP telephones could communicate directly, this would require that one
user knows the IP address of the other user's telephone, an unlikely occurrence.
The call manager is also called a gatekeeper. The call manager provides some or
all of the following services.

Call establishment

Establishing a call requires address resolution. This means that the way that the
originating caller identifies the recipient, by telephone number, name, extension or
IP address must be mapped to that recipient's IP address if the recipient is also
using Voice over IP. On the other hand, if the recipient is on an external telephone
system, the call manager must forward the request to the PSTN access device, the
VoIP gateway.

Call initiation

The call manager then proceeds to initiate the call, by using H.225, SIP, or
proprietary signaling to contact the called party. Upon a successful connection, the
gatekeeper “hands off” the call to the two connected phones or phone-gateway
pair. If the call cannot go through, because the called phone was in use or
otherwise unavailable, the gatekeeper will then inform the caller with a busy signal
or redirect the call to an automated attendant or voice mail system.

Admissions control

The call manager can control access to the telephone system to authorized and
registered endpoints. Devices unknown to the administrator will be disallowed.

Bandwidth control

Connections can be disallowed if the system has no more bandwidth. A percentage


of the bandwidth of the network may be reserved for data or some critical usage.
The call manager may also restrict the number of people participating in a
video/audio conference.

Zone management

The call manager will manage its zone, defined as the endpoint devices that
register with it. This last includes maintaining a real time list of calls in progress in
order to provide a busy signal as required.

Additional services

The call manager provides some of the services that a traditional PBX provides
such as call hold, call transfer, call forwarding, and call waiting.

The call manager can be a dedicated box, be integrated into another box such as a
router or be software sitting on a server.

The call control functions used by the call manager are H.323 and SIP plus some
proprietary protocols such as Cisco's SKINNY protocol.

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Figure 10: Setting up a call

As figure 10 illustrates, when the phone at one end is picked up a connection is


made to the call manager which in turn contacts the second VoIP device. Only
when the call setup is completed do the two endpoint devices communicate with
each other on a peer-to-peer basis.

Call Setup Services

 H.323
 Session Initialization Protocol (SIP)

H.323

H.323 is a set of protocols used by VoIP to establish and manage telephone calls.
H.323 is controlled by the International Telephone Union (ITU).

Not only voice


H.323 is a very broad set of standards that provide for video and data
conferencing as well as audio. The networks that H.323 was designed to work with
do not provide quality of service. Ethernet is the primary example of this type of
network. H.323 is not specific to Ethernet or any network; it will work with them
all. And if the network does have some QoS, bonus.

Figure 11: The H.323 protocol suite

Many protocols

An umbrella
H.323 is actually a framework for multimedia conferencing and therefore uses
other protocols to handle many functions. See figure 11.

 Control and call signals


H.245 — used to negotiate channel usage and capabilities
Q.931 — used for call signaling and call setup
RAS — Registration/Admission/Status is a protocol used to communicate
with a Gatekeeper
 Audio codecs — G.711, G.722, G.723, G.728, G.729
 Video codecs — H.261, H.263
 RTP/RTCP - used for sequencing audio and video packets

H.323 components

H.323 defines four major components for a network-based communications


system: Terminals, Gateways, Gatekeepers, and Multipoint Control Units.

 Terminals on a VoIP system are telephones and PCs equipped with VoIP

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capabilities.
 Gateways are the bridge to the PSTN.
 Gatekeepers act as the central point for all calls within its zone and provides
call control services to registered endpoints. In a previous discussion, the
H.323 Gatekeeper was called the Call Manager. To recap, the functions of
the Gatekeeper include address resolution and mapping, call initiation and
establishment, admissions control, bandwidth control and zone
management.
 Multipoint Control Unit (MCU) supports conferences between three or more
endpoints. Conferencing can be audio, video or data.

Figure 12: H.323 zones

First on the scene, but…

H.323 is a flexible and comprehensive framework for multimedia conferencing,


including VoIP, which has been implemented on millions of end devices since 1996
(V1) and 1998 (V2). It is known to work well and is reliable. However, there is
room for an alternative call management service because of the following features
of H.323.

 H.323 is not dedicated to VoIP and therefore is complicated because of the


extra baggage it carries around for video conferencing.
 H.323 is not very flexible. New features can be added but they must be
backward compatible.
 H.323 is controlled by the ITU, which moves slowly with new technology,
and is slanted toward the traditional PSTN. In the fast paced world of the
Internet, H.323 is evolving slowly.

SIP is the alternative and it was designed with the Internet in mind.

SIP

SIP, the Session Initiation Protocol, is a signaling protocol for Internet


conferencing, telephony, presence, events notification and instant messaging. SIP
was developed by the Internet Engineering Task Force (IETF).

SIP is used for setting up, controlling and tearing down sessions on the Internet.
Sessions include, but are not limited to, Internet telephone calls and multimedia
conferences. SIP is also used for instant messaging and presence. Note that SIP is
designed for managing sessions (connections) whereas H.323 was designed for
multimedia conferencing. VoIP falls within the scope of both.

SIP is a request-response protocol that closely resembles two other Internet


protocols, HTTP and SMTP (the protocols that power the world wide web and
email); consequently, SIP sits comfortably alongside Internet applications. Using
SIP, telephony becomes another web application and integrates easily into other
Internet services.

Figure 13: SIP in action

SIP architecture

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SIP uses the following components:

SIP URL — Every endpoint on the VoIP system has a SIP URL for identification.
The URL for Bob Jones at ABC.com might be:

sip:bobj@abc.com

Notice how much this URL resembles an email address. This is not an accident. If
placed on a web page, clicking the URL will instigate a phone call to that endpoint.

If calling a telephone number on the PSTN, the URL will look like:
SIP:5551212@gateway, where gateway is the name of the machine that acts as
the gateway to the PSTN.

Registration server — The registration server authenticates the user, and adds the
mapping between URL and network address to the location server's database.
When the user agent starts up, the first message it sends is a REGISTRATION.

Location database — The location database maintains the database of name to


location (IP address usually) mappings. The information in the database is usually
acquired from the user agent registrations, but may be acquired in other ways as
well, such as DNS. The database may be queried in various ways, although LDAP
is the most common.

When a user agent wants to connect to a remote SIP endpoint, it queries the
location database in the location server for the contact information.

Proxy server — Proxy servers, as their name suggests, act on behalf of user
agents, routing SIP messages to correct destinations.

Redirect server — A redirect server differs from a proxy server in that it does not
forward messages but simply does a location look-up and returns one (or more)
addresses for the destination and leaves it up to the original user agent to contact
the destination at these addresses directly.

A SIP server will include some or all of the above functions or the functions can be
split between multiple machines.

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