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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
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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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).
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.
H.323 components
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.
SIP is the alternative and it was designed with the Internet in mind.
SIP
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 architecture
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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.
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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