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SipExchange Home |
Terminologies
The following table explains some of the terms used in the SipExchange
documentation throughout this site. |
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| Advanced intelligent
network (AIN) |
Most modern telephone
switches (also called the Service Switching Point or SSP) support the
concept of an "external call control" mechanism. While handling a call
between two parties, on certain pre-configured points (called trigger
points) in the call, it can send a query to an external Service Control
Point (SCP) asking the SCP how the call should be handled. The SCP
sends
a response specifying how the call is to be handled. The responses may
include (a) terminate the call, (2) route the call to another user, (3)
continue processing the call. This mechanism is called the Advanced
intelligent network or AIN. AIN enables SCPs to provide external call
control and enables new features to be developed outside of the SSP.
For
example, if during call setup, the party being called is not available,
the SSP is configured to send a query to the SCP at this point of the
call, and the SCP sends a response forwarding the call to a voice mail
device, the voice mail feature is implemented without having to make
any software changes to the SSP. The idea is that the SSPs are
typically legacy systems that are difficult to add features to and SCPs
are computers systems where such service logic can be created more
easily. There are standards that have been defined by various telecommunications bodies that specify the trigger conditions, communication protocols between the SSP and SCP and even how the service logic can be created. These standards allow SSPs and SCPs from different vendors to talk to each other. However, in reality, many vendors have their own extensions to the standards, and SCPs and SSPs do not always work together seamlessly. |
| Call detail record (CDR) |
Call detail records are
information about a call that was made or received by a switching
system. It contains all the information about the call including
calling and called addresses, call duration, etc. The CDRs are stored
in the switch database and are later retrieved by billing applications
for creating customer invoices, etc. |
| Calling address |
The calling address refers
to the SIP address of the user initiating a call. |
| Called address |
The called address refers
to the SIP address of the user receiving a call. |
| Domain |
In the IP network, each
computer has an IP address and in many cases, the IP address maps to a
name - called the domain name. Examples of domain name include
"cafesip.org" or "mail.cafesip.org". For the SIP networks, the same
concept is used and each SIP user has a SIP URL that consists of an
user name and a domain similar to an email. An example of a SIP URL is
sip:amit@cafesip.org where "amit" is the user name and "cafesip.org" is
the domain name. When you set up the SipExchange server, it allows you
to manage more than one SIP domain. For example, from the
administration user interface, you can create domains "cafesip.org" and
"quik-j.com" and assign subscribers (or users) to each of the domains.
That way, your SIP service can serve more than one domain. Note that if
you deploy the application in a real IP network, the domain names must
resolve to the IP address of your SipExchange server. |
| Feature | Feature refers to
subscriber features like "call forwarding", "call waiting", "voice
mail", etc. When you subscribe to a telephone service, you can select
from a list of features that the service provider offers. There are
features that you do not see as a subscriber. These features apply to
the entire switch or to a domain. For example, the toll free feature in
a
switch allows a switch to translate toll-free numbers into real
telephone numbers by looking up the dialed digits in an external
database. The
SipExchange application is a switch that comes with many in-built
features and enables features developed by other vendors to be "plugged
in" using the AIN/IN mechanism. |
| Intelligent network (IN) | Same as AIN. These terms
are used interchangeably. |
| Payload |
Payload refers to the
voice, video or other multi-media packets that are exchanged between
telephones to transmit voice/video and other multi-media data. When an
user speaks on a phone, the telephone digitizes the voice into packets
and transmits the packets to the other end. The majority of messages
exchanged between the telephones are payload data because voice, video
or other types of payload tend to take up a lot of bandwidth. An
important requirement is that the payload be delivered in real-time, as
interactive conversations between two parties require a party to
hear the other speak in real-time. |
| Service control point (SCP) | Please read the terminology section on AIN. |
| Service switching point (SSP) | Please read the terminology section on AIN. |
| Signaling |
Signaling refers to
the messages exchanged between telephone and switches or between
switches
to set up and tear-down calls. It also refers to messages exchanged for
call control purposes. The traditional signaling protocols include SS7,
ISDN,
etc. SIP is a popular protocol for signaling used for voice over IP
(VOIP)
calls. Another signalling protocol used for VOIP is H.323. |
| Soft phones |
Soft phones are computer
software that can be installed on your desktop, your PDA or your mobile
phone that allows you to make phone calls. Basically, you can use your
desktop, PDA or mobile device connected to the Internet as a phone.
Examples of such phones include Microsoft Messanger, Skype, Yahoo, etc.
Some of these phones use standard communications protocols like SIP.
Others use proprietary communications protocols. |
| Subscriber | A subscriber is a user of
a service. When you buy a telephone service from a service provider
like Vonage or British Telecom, these companies refer to you as a
subscriber. |
| Trigger |
Please read the terminology section on AIN. |
| Voice over IP (VoIP) |
Initially, the Internet
was envisioned to provide data services like email and the world wide
web. Then came the idea that we can also use the Internet network for
setting up voice and video sessions. In other words, we can use the
Internet for making phone calls for free! For this to work,
communicatons standards must be defined so that phones and servers from
different vendors can still understand each other. There are a number
of competing standards for VoIP communications. H.323 and SIP are some
of the most popular ones. In recent years, SIP has emerged as a
protocol of choice. |
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