GSM architecture : Base Station Subsystem (BSS) (2/3)

GSM architecture : Base Station Subsystem (BSS)

Several BTSs are controlled by the Base Station Controller, or BSC.
This assigns free radio channels in the TRX for the link to the mobile station. It controls the
necessary output power for mobile station and TRX. It monitors the existing radio link to and
from the mobile station and controls handover between neighboring radio cells if they are
under its control. During an existing radio connection, the BSC monitors its quality and
controls disconnection of the radio link when the call is over. The BSC communicates with
the transcoder (TC) via the A-ter interface.

GSM architecture : Base Station Subsystem (BSS) (1/3)

GSM architecture : Base Station Subsystem (BSS)

The Base Station Subsystem ensures as complete a network coverage as possible and
includes a large number of structurally organised radio cells. It consists of the following
elements:
• The Base Transceiver Station
• The Base Station Controller
and
• The Transcoder

The central element of one cell of this kind is a transmitting and receiving unit known as a
Base Transceiver Station (BTS). This makes the connection to the mobile station via the air
interface and controls the transceiver (TRX). The transceiver, the central functional unit of
the BTS, maintains calls to a maximum of 8 mobile stations via one frequency pair each. The
BTS is also responsible for the monitoring of the signal quality and the encoding and
modulation of useful signals. Via the A-bis interface, it forwards calls, signals and control
information destined for the OMS and the NSS to the Base Station Controller (BSC).

GSM architecture

GSM Architecture :Network Elements and their Basic Functions

For the subscriber, a mobile telephone call is a simple process. In reality, though, this call is
only possible thanks to a complex network architecture consisting of various different
network elements. In this lesson, you' ll get to know the individual elements of the GSM
network and their basic functions.

gsm architecture

The Base Station Subsystem BSS provides the connection between the mobile stations and
the Network Subsystem NSS. The NSS forwards user signals to other mobiles via the BSS
or subscribers in the Public Switched Telephone Network (PSTN), and provides necessary
customer data. The Operation & Maintenance Subsystem (OMS) monitors BSS and NSS
performance, and remotely debugs occurring faults in the network elements.

Additional components such as interface elements to data networks, the Short Message
Service Center or the Voice Mail System complete the GSM system architecture.

Echo Cancellation

Echo Cancellation

An echo might be amusing enough when we are hiking in the mountains, but it is certainly a

nuisance during a phone call and might even make communication impossible.

Hearing one's own voice within the handset is a normal effect that calms the speaker. But a

delay between the direct sound and the echo of more than 25 ms might cause interruptions

and disturb severely the rhythm of communication.

Legacy PSTNs have to cope with echoes produced by non-aligned impedances when the

signal leaves the 4-wire system of the switch to enter the local 2-wire trunk to the end user.

Very careful impedance supervision at the reflection points and the use of echo cancellers

helps to solve the problem.

In VoIP systems, echo cancellation is included with the codecs in the DSPs or digital signal

processors. Hence, a software-based solution is used for IP soft-phones and the gateways of

the network. For a certain period of time, it stores an inverse pattern of the sent speech

sample. Then the DSP listens to the sound that is reflected by the distant party and subtracts

the stored signal. This achieves a near-zero amplitude for the sound disturbance. The

majority of echo cancellers cope with 32 ms echoes. That is the maximum storage time for

the speech sample.

Voice Activity Detection

Voice Activity Detection

In a phone call, normally one party speaks while the other one listens to the speaker. As a
standard PSTN link provides 64 kbps full duplex channels, half of the capacity is usually
wasted during this kind of communication.

Voice over IP makes these resources available for other purposes if Voice Activity Detection
is activated.

How does this work? VAD detects the loudness of the speaker's voice and decides when to
stop speech frame packetizing. Before doing so, VAD waits for a fixed period of time, mostly
200 ms. In very noisy environments, VAD might have problems distinguishing between
speech and background noise. At the start of the call a defined signal-to-noise ratio, also
called the signal threshold, is used to decide whether to automatically activate or de-activate
the VAD operations.

If the VAD procedure detects the absence of voice and simply switches off speech
information transmission to the distant party, they might think that the call has been
interrupted. In order to avoid this, the systems inserts an artificial noise, called Comfort
Noise, to make the listening party believe that the link is still present.

Codec Bandwidth and Delay

Codec Bandwidth and Delay

When we are looking for the most efficient use of bandwidth, the aim should be to pack the
maximum number of speech samples into each single IP packet. However, this approach is
limited by the available network connection speed, which might require smaller data blocks.
This is shown clearly in the following example. On a 28.8 kbps modem-type trunk, an 86 byte
data packet needs 23.8 ms for transmission whereas on a 100 Mbps 100Base-T Ethernet
link only 6.8 μs are needed.

In both scenarios, additional time is needed for packetizing as the IP packet has to wait until
the last speech sample has been encapsulated before network transmission can start. For
example, including 3 additional speech frames in the same IP packet will produce a waiting
time of 30 ms with 10 ms more waiting time for every further speech sample.
Real-time IP applications, such as VoIP, are very delay sensitive. All delays - that is, speech
coding and compression, packetizing and transmission - have to be countered to keep the
overall delay time at a level where speech quality is acceptable.

Codec and Bandwidth (2/2)

Codec and Bandwidth

This seems quite bandwidth-inefficient as each subsequent 10 byte speech payload comes
with a dedicated overhead. To solve the problem, more than one speech frame needs be
transmitted per IP packet.

How can we increase the payload per Ethernet frame?
Let's try to pack 2 frames of speech information at 10 ms coming out of a G.729 codec and
calculate the resulting bandwidth. 66 bytes of overhead are responsible for 2 x 10 ms of
payload hence 86 bytes per frame or 688 bits. As two 10 ms speech frames are carried the
resulting amount of packets per second has to be divided by 2. This gives us 50 packets.
50 packets per second at 688 bits correspond to a bit rate of 34 kbps or 68 kbps for the full
duplex line capacity. Thus, a significant bandwidth reduction of 50% has been achieved.