Continued....
In the UK video from most sources
runs under the PAL standard which 25 frames per second – e.g.
domestic VHS players. In the USA the system is NTSC which runs at
30 frames per second. Videoconference equipment commonly is
specified as 15 fps or 30 fps. In layman’s terms - 15 fps will
be jerky, it shows every other frame, 30 fps is full motion video.
The higher the frame rate the
greater the data load, so the faster the connection required.
A minimum requirement for full
motion video and audio between two points is 768 Kbps.
This made up of:- an encoding rate
of 384 Kbps (typically used) is selected. This is broken down into
two parts –
- 64Kbps is for the audio
- 320 Kbps is for the video.
The resulting 384 Kbps stream is
compressed and sent (from you – the source) to the remote point
(the destination). Similarly a 384 Kbps stream is received from
the remote point to you. Thus twice 384 Kbps in bandwidth is
required.
If there is a lot of motion in the
video, very little compression is achieved. If there is little or
no motion in the video, the savings can approach 50%. It would,
however, be foolish to design a system which relied on the
participants being forced to remain static.
There are two types of video
conferencing, each has sub-types.
- Point to point - a live video /
audio communication between any two locations.
- Multipoint - links between a
three or more locations
Point to Point (P2P)
Point to point - a live video and
audio communication link between any two locations.
P2P - Application
Where the need is to communicate
between two points only at any one time:
- Two offices of the same company
- Yourself and a business partner
company, e.g. Yourselves and a major supplier
- Senior management/ teams from
two divisions – e.g. research and manufacture.
P2P - Use
- Virtual meetings on a one to one
basis
- Project workgroup co-ordination
of effort, live adjustment of data, drawings, documents or
prototypes
- Virtual board meeting between
two groups of people in specific locations - say 5 in one and
4 in another
P2P - Limitations
- Bandwidth required increases in
proportion to the amount of data being exchanged.
- Audit trail of actions taken by
whom with a timeline sequence often required.
- Extra equipment is necessary to
show physical objects, using a visulaiser for say engineering
parts
Point-to-Point Videoconferencing
Consider two videoconference
terminals (vct) that are connected to the Internet.
The vct and its associated
peripherals allow the user to make a call to another client, send
the local audio/video stream to the remote client, and hear/view
the received audio/video stream on a local speaker/monitor that is
connected to the vct. Assume one user (the local user) uses a vct
to call a user at a remote vct by entering the IP address of the
remote vct. The clients setup a call between the stations
following the specifications of the H.323 protocol. Once the call
is setup, the clients exchange audio/video streams over the
Internet. The point-to-point videoconference continues until one
of the users "hangs up" the call.
IP numbers are difficult to
remember; some users have dynamically assigned (DHCP) IP numbers
that can change every time they boot their system and problems in
using IP addressing when different vendor systems are used.
The Gatekeeper
To alleviate the problem of IP
dialing, the H.323 standard defines the use of a gatekeeper.
The gatekeeper is a system that
connects to the Internet just like the client terminals. The IP
address of the gatekeeper is configured into the client terminals
and when the clients "power up", they communicate with
the gatekeeper and transfer certain information to the gatekeeper
that describes the vct.
When the clients register with the
gatekeeper, they pass their IP numbers, H.323 alias, and H.323
extension to the gatekeeper where it is stored. This allows a
local user to dial a remote user by entering the remote users
H.323 extension in effect their video telephone number.
The local vct communicates the
H.323 extension to the gatekeeper. The gatekeeper then checks to
see if the remote client is registered with the gatekeeper, then
sets up the call between the two clients.
Once the call has been setup, the
audio/video streams flow directly between the clients over the
Internet.
Multipoint
Live video and audio links between
a three or more locations. To handle this situation, the H.323
standard introduces the concept of a Multipoint Control Unit (MCU).
The MCU can be thought of as a "video bridge". The MCU
connects to the Internet and registers with the gatekeeper.
A MCU, depending on its design
capacity, can handle a certain number of simultaneous
videoconferences each with each videoconference being logically
separate from the others and with each having a specified number
of users.
When users want to join a
particular videoconferencing session, they dial the service
number/password combination. The gatekeeper checks to see if that
service has been registered by a MCU. The gatekeeper completes the
call by connecting the client to the specified videoconference on
the MCU.
Once the call has been connected,
the client's audio/video stream is then sent over the Internet
from the client to the MCU. Similarly, other clients connect to
the session and send their audio/video streams to the MCU. The MCU
selects one of the audio/video streams on the videoconference and
returns that audio/video stream to all of the clients (that is all
except the client whose stream was selected).
There are several methods for
selecting an audio/video stream. Audio switching and chairman
control are two alternatives. Typically, the method that is chosen
is audio switching where the MCU selects the stream that currently
has active audio (someone is talking or is talking the loudest).
As the user(s) at one site stop
talking and the user(s) at another site start to talk, they
capture the MCU. The process is repeated with the video from the
newly selected site now being sent to all the other sites.
Streaming
To participate in a H.323
videoconference, users must have appropriate videoconferencing
client terminals and have Internet connectivity with sufficient
bandwidth to support the videoconference.
Some users may not have these
capabilities but would still like to be able to participate even
if that meant that they could only see and hear conference
participants but not be able to interact with them. (Watch and
Listen)
Users can receive the stream using
a browser on a computer. They enter the URL of the server, and the
server starts the encoded audio/video stream over the Internet to
the computer. Plug-Ins for the browser exist that are capable of
decoding both RealVideo and Windows media streams. The user can
thus see and hear the participants in the streamed videoconference
in near real-time.
Alternatively, a user can connect
to the server at a latter date and view the archived version of
the videoconference.
About the
author:
James Hunter
works for Edric Audio Visual, one of the premier suppliers of video
conferencing in the UK.