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Beginners section
What is ISDN?
ISDN stands for Integrated Services Digital
Network and is, put simply, is a digital equivalent to a telephone
line. ISDN services are available from most telecomms operators
in most parts of the world. What is descibed as 'an ISDN
line' is actually made up of two different types of channels.
D-channels are used for signalling and call setup. B-channels
are the data channels and are used for the transfer of your data
(and voice calls - ISDN lines can still carry those too!)
Each B-channel can be used independently as a separate data channel
(or telephone line), so you can have two separate connections
(or calls) in progress at the same time.
Typically ISDN is available in two seperate
forms. Basic rate ISDN (also called BRI or Basic Rate Interface
or ISDN-2 in Europe) consists of two B-channels and one D-channels.
ISDN audio codecs are generally connected to BRIs.
The second type of ISDN service is called Primary Rate Interface
(PRI or ISDN-30 in Europe). This is used by bigger installations
as it provides one D-channel and up to 30 B-channels. Devices
designed for the BRI will not connect directly to PRI installations
- but will connect through the ISDN PABX's normally used
with primary rate ISDN.
ISDN offers several key advantages over standard
analogue phone-lines. It has a higher bandwidth, so data
can be sent faster. It normally provides better call quality
as the line is digital right throught o the point it reaches you.
ISDN lines can also use Multiple Subscriber Numbering (MSN's)
to allow you to have several devices attached to your lines, each
with their own unique telephone number. So, for example,
you could have five different telephones attached to a single
ISDN BRI. You still only have 2 B-channels, so only two
can be in use at once, but all five could have their own phone
number.
One current disadvantage to ISDN is that
existing telephones, fax machines, modems etc. cannot be connected
directly to it. These Plain Old Telephony devices (POTS)
need a Terminal adapter between them and the ISDN line.
ISDN audio codecs and other devices, which are designed for ISDN
do connect directly (they have their own TA built in).
A single B-channel (and remember you have
two, even with a BRI) can carry 64kbits of data per second -
that's twice as much as an analogue phone line. ISDN lines
are used for many different applications including:
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Broadcast and Professional Audio (ISDN
audio codecs)
-
Newspapers - for rapid worldwide
distribution of photographs and stories
-
Internet Access - up to four times
the speed compared with connecting through an analogue phone
line
-
Video-conferencing and PC based desktop
versions
-
LANs for homeworking and interconnection
or backup of computer networks
-
Group 4 fax - which allows you
to send a sheet of A4 in laser quality in seconds.
What
is a codec and how do they work?
Whilst we've said above that you can use
ISDN lines for ordinary telephone calls, and that the quality
is better, the audio will still sound like a telephone call.
Typically ISDN telephony provides a bandwidth of 3.1kHz.
You can calculate the bandwidth you would
need to carry live stereo CD quality audio as follows: CD
quality is two channels of 16-bit audio send at a sample rate
of 44.1kHz. This comes to 1.4 mbits per second. Much
more than the 128 kbits you have with the two B-channels available
on BRI ISDN. The codec's function is to compress and un-compress
(or COde and DECode) the 1.4 mbits of data into the 128 kbits
of space available over the ISDN line.
What
kind of quality can I expect?
The manufacturers make many different claims
here! Basically, it depends on the way the audio is compressed
- but remember that it is just that - compressed
audio; there's nothing cleverer than that going on.
With a single Basic rate (ISDN-2) ISDN installation,
and using both of the B-channels at the same time, you can expect
to get FM broadcast quality stereo audio. Using a single
B-channel (either by choice, or because some cheaper codecs are
only capable of using one channel for each connection), you will
still get near broadcast quality speech. This latter mode
is often used for down-the-line interviews or commentary from
sports matches etc.
Some codecs (like the Musicam range) allow
you to use up to six B-channels simultaneously. Of course
this assumes that you have 3 sets of ISDN lines available.
Using all six together allows a data rate of 384 kbits to be achieved.
This means that the audio is less compressed, and so better
quality. Bear in mind that using any more than two B-channels
at once does add significantly to the transmission delay of your
audio.
How
do I choose the correct manufacturer or model for me?
Take a look at our product range, and also
ask your broadcast colleagues what they would recommend.
If you need a system that will connect with other people's, read
the section on interoperability below.
What
are the various protocols used?
There are two types of protocol for you to
consider. The first is the telecomms protocol. The
second is the audio coding protocol. These are explained
below.
Main Section
ISDN
Worldwide Protocols & interoperability
ISDN is not quite the same all over the world.
Telecomms providers in different countries (and in the US, in
different states) may use slightly different ISDN protocols.
When you buy a codec, you buy it with the correct type of Terminal
Adapter installed. Some TAs can be switched to work with
different protocols, others are fixed to one or a selection of
them.
This is only a case of you having the right
equipment to connect to your line (rather like having a 110v or
220v toaster - you have to have the right power supply)
it does not affect your ability to connect to other people
who's equipment may be connected to another protocol
in another country.
Having said that: there is one exception
to this rule. In some parts of the states, there is still
an older version of ISDN in place, known as Switch56. This
offers B-channels that can only carry 56kbits each, rather than
the usual 64kbits. Musicam codecs will allow you to connect,
with a little hope and prayer, to codecs on Switch56 lines.
Others normally don't. One note on Switch56: it's
fair to say that a line that carries 14% less data will offer
significantly poorer audio quality than 64k standard ISDN, hence
few pro-audio users do actually continue to use this old system
in any case.
Audio
protocols and interoperability
There are four main methods or protocols
which different types of codec use to COde and DECode the audio.
They are: MPEG Layer II (MPEG-2), MPEG Layer III (MPEG-3),
APTx and G.722. For you to make a successful audio connection,
both codecs must be using the same method. Most codecs are
capable of using one or two. Only Glensound make codecs
which are capable of three of the above.
G.722 by a cat's whisker, is the most universally
compatible of all the audio coding protocols. MPEG-2 is the
most widely used for broadcast audio, offering compatibility and
quality.
G.722 uses just one B-channel to send decent
quality speech over ISDN. Baiscally, if two codecs are capable
of using G.722 they will always work together.
MPEG Layer II codec interoperability can
be more complex. In theory they should all be inter-compatible,
but... For example, Musicam codecs, although they are fully MPEG-2
compatible, need to be set in a special mode called 'decoder independant'
before they will connect to other MPEG-2 codecs. The ISYS
Pro system is the only codec currently available with an 'auto-detect'
algorithm built in. This queries any other codec with which
you try and make a connection and automatically sets itself to
'speak the right language'. ISYS Pro is quite new to the
marketplace, and this feature has given it a real edge.
MPEG Layer II and MPEG Layer III are both
open coding standards (i.e. manufacturers don't need a license
to use them) designed by the Motion Picture Experts Group -
so called because they also define a standard for video compression.
MPEG Layer II is the best method for broadcast quality
audio, and works over one or two B-channels. FM broadcast
quality over two channels, speech quality over just one.
MPEG Layer III is a newer standard developed by this group, which
is in theory much better than MPEG-2. However, most professional
audio users still prefer MPEG-2 - see What about MPEG Layer
III - later in this document.
APTx is something of a Betamax amongst coding
standards. Although it offers several key advantages over
the other standards mentioned above, it's not one to go for if
easy connection to other people's codecs is important to you.
What is Musicam?
It's Musicam USA's own version of MPEG-2
compression.
MPEG-2 is a standard defined by committee.
However, it's possible to change the performance of an encoder
by changing the parameters it uses is analysing the signal.
A signal produced in this way is still decodable by other manufacturers'
codecs.
As you'd expect, all manufacturers claim
that their encoder works better than anyone else's. Musicam
have gone one stage further by giving their version of MPEG-2
its own name.
Philips are one of the biggest companies
on the MPEG-2 committee; their own commercial MPEG-2 coding
technology is used in ISYS Pro.
What about MPEG
Layer III
This is the newest coding method in the marketplace
and is supported by a couple of manufacturers including Musicam
USA.
It is a far better way of coding the signal.
However, many broadcast and pro-audio users prefer Layer II for
the following reasons:
-
MPEG-2 offers a much shorter transmission
(coding) delay than MPEG-3. This is because one of
the things that makes MPEG-3 better is that it looks further
forward at what's about to happen in the audio than MPEG-2.
As a consequence, it's impossible to design an MPEG-3 codec
with as short a delay as MPEG-2 systems.
-
Whilst MPEG-3 is a better technique for
some types of audio, when it gets it wrong, it gets it more
wrong! Let us try to explain. All codecs compress
the signal. Experienced ears can hear the imperfections
in any compressed audio. The most widespread use of
audio codecs is in radio. The imperfections in MPEG-3
audio, unfortunately, are worst in speech or audio including
speech. Many audio professionals will say that these
imperfections are, in general, less audible in MPEG-2, whether
for music or speech.
-
Most of the installed-base of codecs
around the world are capable of Layer II. A far smaller
number of Layer III codecs have currently been installed.
-
MPEG-2 is the new standard which has
been adopted for DAB (digital radio broadcasts), DVD and Digital
television sound. As a result, most of the new R&D
money is going into further developing MPEG-2 not MPEG-3.
some manufacturers including Philips already boast that their
MPEG-2 encoders offer better quality than the newer MPEG-3
devices.
-
Audio that has been passed over an MPEG-3
link sounds absolutely appalling (near un-broadcastable) if
then transmitted or carried in a studio-transmmitter link
over MPEG-2. For this reason alone, pro-audio users,
particularly in broadcast avoid MPEG-3 ISDN codecs.
I'm
trying to connect to someone with a ....... codec and it doesn't
work!
This really is one to try posting on the
discussion board! But, in brief:
Make sure that the codecs are compatible
- do they both have an audio coding protocol in common?
Try setting each codec manually to use exactly
the same bit rate, sample rate, mono/stereo mode and audio coding
protocol. i.e. over-ride the 'auto' settings.
Try a different protocol. The two almost
universal ones are: MPEG Layer II at 64 kbits mono and
G.722. If neither of these work, there's something very
wrong.
If this is a regular problem for you, then
you should take a look at our page on the ISYS
Pro system, which features a unique 'auto-detect' algorithm.
As far as we've seen it in operation, if it's going to connect,
ISYS Pro tends to get it right first time very nearly all of the
time.
Live connections
and coding delays
The shortest possible codec delay is obtained
using G.722 or APTx. MPEG-2 delays at FM broadcast quality
are around 200ms each way. ISYS Pro reduces that delay by
around half. However, the really clever thing to do is to
split-code. That is, you send 'cue' via G.722 and receive
the audio via the higher-quality MPEG-2. The very latest
software update enables the Musicam codecs to do this. ISYS
Pro also offers this functionality.
Sending audio to
multiple locations
The Musicam codecs really come into their
own on this one. A fully equiped CDQ Prima 2xx with three
internal TAs installed is capable of sending a speech quality
signal to six separate destinations simultaneously!
Other units definitely worth a look for this
type of application include the Glensound range. They have
a range of 2-inch wide 3U high codec modules - they're so
tiny that you can get over 20 of these into 3U of rack-space.
They also have a 1U rack system into which you can install
two independent codecs/TAs.
Installing
a single codec for use in multiple studios
Let's say you have one ISDN audio codec which
is used in a transmission suite for part of the day, and also
in a production suite. If the codec only has to answer incoming
calls, they you can simply take a spare feed of the audio send
& return.
However, most of the time you also need to
be able to use the front panel of the codec to dial & make
settings. So, there are two considerations here. Firstly
you have to be able to route the audio feeds. Secondly you
have to somehow route the control interface of the codec(s).
Assuming that you can take care of the audio
yourself! let's move on to the control interface.
CDQ Primas are remote controllable via an RS-232 interface, so
you could install a custom script in your computer network server
to control a Prima. We also sell a Windows application which
can remote control a CDQ Prima from a PC, and on request, we can
supply (for free) software that extends this functionality over
a PC network.
ISYS Pro comes ready for this application
- new and existing users should simply ask us for the remote control
package which will allow you to use your ISYS system (or multiple
systems) over your PC network. The host and guest PCs have 'dual-control'
of one or more systems. This is a very elegant solution
for broadcast control rooms and comms areas.
That's the end of the current FAQs. If you
didn't find the answer you were looking for here, try our discussion
message board, where one of the moderators or another user
may be able to help! If your enquiry is a pre-sales question,
e-mail us at enquiries@isdnaudio.com
We'll continue to update this FAQ with the most frequent
queries on this board.
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