In this era of convergence, sending voice over networks–like IP, ATM and
frame relay–has become the hottest technology facilitating real time
communication. The basic element in the technology of Voice over Internet
Protocol (VoIP) is the packet. It uses the packet switching network to transfer
voice across the network. But the challenge that lies before VoIP technology is
that it has to route its service through the traditional circuit-switching
network. To address this issue, the International Telecommunication Union (ITU)
has defined a standard–the H.323 for packet based multimedia networks.
A circuit switched network or the public switched telephone network creates a
dedicated path between the original and the destination call. Although the same
two-end points of the call can follow different paths, the circuit is fully
available for the duration of the call, making it unavailable to other network
users. But, in the packet switched networks there is no dedicated path between
the users. Here the content is broken up into packets and sent across the
network. Each packet carries a header, which determines the destination of the
packet. Here too the packets take different paths towards the destination but
they share their paths with other packets.
Now let’s see how H.323 works in this process. They are connected to the
ISDN, PSTN and your local landline or wireless devices. The components of H.323
are:
- H.323 terminals which are end points to the LAN
- Gateways
- Gatekeepers
- Multipoint control units.
The H.323 are the LAN based end point terminals. These terminals include the
Codecs and accomplish the voice transmission functions. Codecs are
compression/decompression devices that encode or decode a signal. When the
signal is determined to be voice, it is usually compressed by a DSP from 64K PCM
to one of the compression/decompression (Codec) format. The voice packet is
constructed as an IP packet, to avoid TCP/IP to attempt to correct a corrupted
packet by re-transmitting the packet. Any attempt to retransmit the packet would
result in unnecessary delays. They are hardware or software components providing
digital encoding and decoding of analog signals. They send and receive
packetized voice. These terminals also need to support signaling functions that
are used for call setup. The terminals are connected to the gatekeepers through
their RAS (Registration, Admission, and Status) point.
Gateways are interfaces between the LAN and switched circuit network. The
gateway serves as the interface between the two different physical networks. It
forms a connection between the packet switched network and the traditional
circuit switched network. It performs the task of translating the signaling
messages between the two sides and also compresses and decompresses the voice.
The originating gateway establishes connection to the destination gateway,
exchanges call set up and compatibility information, and performs any
negotiation and security handshake. At the same time, the gateway is also
receiving packets from other IP gateways and decompressing voice information
back to the original format, to be connected to the appropriate telephony
interface. When the IP gateway is required to place a call, it receives a called
party phone number from the calling phone and converts it to the IP address of
the far-end or the called party’s gateway (destination gateway).
Gatekeepers are usually present to perform a set of functions like provide
address translation (routing) for devices in their zone. The translation can be
between external and internal numbering systems. They also specify what devises
can call what numbers thus providing admission control. A gatekeeper can take
part in a variety of signaling models as commanded by the gatekeeper. The
signaling models determine what signaling messages pass through the gatekeeper,
and what can be passed directly between entities such as the terminal and the
gateway. There are two such kinds of signaling models. A direct signaling model
does not involve a gatekeeper for exchange of signaling messages, while in a
gatekeeper routed call signaling model, all signaling passes through the
gatekeeper.
The Multipoint Control Units can be standalone devices, or be integrated into
a terminal, a gateway or a gatekeeper. They have two parts, each performing
certain functions. The Multipoint controller handles the signaling and control
messages essential to set up and manage conferences. The Multipoint processor
accepts messages from endpoints, replicates them and forwards them to the
correct participating points.
But the most important aspect of the technology is the voice quality. This
can be affected primarily due to the type of Codec used and also because of
latency, jitter and packet loss.
But the technology enables the transfer of your voice in the form of packets
through a traditional circuit-switching network to its right destination.