Transcript Signalling System 7 (SS7)

The Higher Institute of Industry
Postgraduate Program
Next Generation Networks (NGN)
Course Instructor
Dr. Majdi Ali Ashibani
Email: [email protected]
Next Generation Networks (NGN)
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Introduction
PSTN
Mobile IP
GPRS/UMTS
4G mobile networks
VOIP
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VOIP QoS issues
Multimedia Control Protocols
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course agenda
H323
H324
Session Initiation Protocol (SIP)
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Soft Switching
Convergent Networks
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Service Delivery Platforms (SDP)
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IP Multimedia Subsystem (IMS)
OSA/Parlay
Next Generation Billing Systems
Ad Hoc Networks
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Ad hoc routing issues
Ad hoc network security
Network Realities
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In the last 10 years have seen an increasingly fast integration
between computers and telephony, both equipment and networks.
Traditional Telephone operators have seen a decreasing in their
revenues from PSTN, due in part to:
 The increasing popularity of mobile telephones
 Movements of communication services to the public Internet.
Customer's demands have moved from one regulated network
provider to unregulated huge content and communication
possibilities offered by the Internet.
This demand have been met by fixed network operators by
deploying broadband network access.
While this solution satisfies customer demands, but network
operator is left merely providing access to public Internet while
content and services been provided by application providers.
Customers buy services and not technology.
The objective is to offer services that can take advantage of
broadband.
Network Realities
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Different operators in different countries with different
constraints will have different views:
Individual networks optimized for the service
Integrated network easier to manage
Integrated network easier for customer (one NT)
Change must be evolution, not revolution: customers
and Operators will not change out equipment
overnight
Customer wants to buy services not the network
Successful network sells the services the customer
wants at the price he/she is willing to pay.
Network Realities
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The concept of a new, integrated broadband
network has developed over the last few years
and has been labeled as next new-generation
network (NGN).
NGN can be determined from the problem faced
by the network operators:
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The need to provide services over broadband accesses
to increase revenue
The need to merge diverse network services, such as
data (web browsing), voice, telephony, multimedia, and
emerging new Internet services.
21st century network needs a series of networks that
could support a flexible platform for service delivery.
Network Realities
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One of the most important feature of IP is
independent of protocol layers (upper and lower).
This feature has greatly impacted global
connectivity networks, which provide IP
connections independent of any kind of network
such as PSTN, ATM, Frame Relay,…etc.
A NGN therefore aims to combine the best of
both worlds from the PSTN and the Internet.
NGN Requirements What
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is an NGN?
NGN: is a packet based network able to provide
telecommunication services and able to make use of multiple
broadband QoS-enabled transport technologies and in which
service-related functions are independent from underlying
transport-related technologies.
A NGN standards process have been started since 2003.
The ITU-T Study Group 13 defined an NGN in
Recommendation Y.2001as:
 A packet-based network able to provide telecommunication
services and able to make use of multiple broadband, QoSenabled transport technologies, and in which service-related
functions are independent from underlying transport-related
technologies.
 It supports generalized mobility which allow consistent and
ubiquitous provision of services to users.
NGN Definition: Recommendation Y.2001
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Packet-based transfer
Separation of control functions among bearer capabilities, call/session, and
application/ service
Decoupling of service provision from transport.
Support for a wide range of service, applications and mechanisms based on
service building blocks.
Broadband capabilities with end-to-end QoS.
Interworking with legacy networks via open interfaces.
Generalized mobility
Unrestricted access by users to different service providers.
A variety of identification schemes
Unified service characteristics for the same service as perceived by the user
Converged services between fixed/mobile
Independence of service-related functions from underlying transport
technologies.
Support of last mile technologies.
Compliance with all regulatory requirements, for example concerning
emergency communications ,security, privacy, and lawful interception.
NGN Definition: Recommendation Y.2001
NGN arrears that has to be addressed by ITU-T and standards
development organizations (SDOs):
 General framework and architectural principles
 Service capabilities and service architecture
 Interoperability of services and network in the NGN.
 Telecommunications capabilities for disaster relief.
 Architecture models for the NGN
 End-to-end QoS
 Service platforms
 Network management
 Security
 Generalized mobility
 Network control architecture (s) and protocols
 Numbering, naming, and addressing
NGN Definition: Recommendation Y.2001
NGN is still under developing, generalized mobility
introduces:
 Fixed-mobile convergence FMC
 Decoupling of service provision from transport
 Provision of open interfaces adds the complexity of
adapting public Internet approaches to provide the
same safe, secure, and reliable networking as does
the PSTN.
 NGN attend to replace PSTN
NGN Architectural Concept
Applications
Other Multimedia
Streaming
services
Access
Network Attachment
Subsystem
IP Multimedia
Subsystem
Resource Control
Subsystem
Access
Transport
IP
Network
3GPP IP-CAN
PSTN/ISDN
Emulation
Subsystem
Core Transport Network
Other Networks
)
(e.g.
PSTN)
Applications and Services
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With increasing complexity of services and a variety
of networks it is important to distinguish between
applications and services:
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A service is what the customer pays for
An application is how a customer makes use of a service
A teleservice is one of standardized set of services
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For instance, a terminal session could make use of a
dialup PSS (X28) service or of the PSTN service.
Switching in PSTN
Switching in PSTN
Switching in PSTN
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In PSTN subscribers are connected in star topology.
Every subscriber is connected to another via at least
one if not many hubs known as offices.
These offices called switches.
Local offices are used for local service connection
and they are known as central offices and use class 5
switches.
Tandem offices used for long distance service and
they use class 4 switches.
A large city might have several central offices.
Mix of topology is used in PSTN.
Tandem Switching Example
Network Connections
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Goal: Each subscriber can reach any other
within the network
Network Connections:
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Mesh
Star
Double Star tandem exchange
Higher-order Star
tandem exchange
Network Topologies
Switching in PSTN
mix of topology in PSTN
Switching in PSTN: links for 1-level hierarchy
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Example: If a Libyan populated area approximately 200000 km2
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assume average service area of a LEX is a circle of radius 3km, an
area of ≈π x 9 km2
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this leads to a requirement of 7070 local exchanges and
24988915links if they are fully interconnected
Switching in PSTN: links for 2-level hierarchy
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Assume number of trunk exchanges (class 4 switches)
is 100.
if trunk exchanges are fully interconnected they need
100×99/2 = 4950 links
Each local exchange has 1 link to the trunk exchange
another 7070.
The total number of links is now 12020 - much less than
before.
Switching in PSTN: links for 2-level hierarchy
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Switches avoid need to fully interconnect
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Allocation of resources
Path set up
Switching in PSTN: links for 2-level hierarchy
Signaling System 7
Signaling System 7
Tandem Switch
Signaling System 7
Dual Tone
signaling
Access Network
POTS
Q931 ISDN
PBX
Switching in PSTN
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Figure 2.2 page 11 Soft switching
Switching in PSTN
Class 4 & 5 switches are the “brains” of the PSTN.
Class 4 Switch
Misurata
Class 5 Switch
Misurata
Class 4 Switch Class 4 Switch
Khoms
Tripoli
Class 5 Switch
Tripoli
Switching in PSTN
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A Class 5 switch is a Local switch capable of
connecting with other class 5 offices within a calling
area; a class 5 switch does not have to have a
trunk to every other class 5 switch in a calling area
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A Class 4 switch (tandem switch) is a Toll switch
that allows the aggregation of several facilities to
connect to other class 4 offices. For routing over
long-distance network.
Compression
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Compression of speech, audio and video relies on the
nature of these signals by modeling their generation
mechanism and/or exploitating human perception limits
(e.g., audible spectrum, tone masking, spatiotemprol visual
filters).
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Entropy reduction and entropy coding:
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Redundancy Reduction
Voice Digitization
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One of the first processes in the transmission of all types of
information that can be captured by human perception, and can be
handled by human made telephone call is the conversion of an
analog signal into a digital one.
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This process is called pulse code modulation (PCM)
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PCM is a four step process consisting of:
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Pulse Amplitude Modulation (PAM) Sampling.
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Companding
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Quantization
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encoding.
Voice Digitization
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Pulse Amplitude Modulation (PAM): the first step in
PCM is known as PAM.
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To represent an analog signal as digitally encoded bitstream.
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The analog signal must be sampled at a rate that is equal to
twice the bandwidth of the channel over which the signal is to
be transmitted.
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As each analog voice channel is allocated 4KHz of bandwidth,
each voice signal is sampled at twice that BW or 8000
sample/sec.
Voice Digitization
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Companding: The second process of PCM is
companding.
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It is the processing of compressing the values of
the PAM samples to fit the nonlinerar quantization
scale. It results a BW savings of more than 30%.
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It is called companding as the sample is
compressed for transmission and expanded for
reception.
Voice Digitization
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Quantization: Third stage in PCM is quantization.
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In quantization, values are assigned to each sample within a
constrained range.
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The difference between the actual level of the input analog
signal and the digitized representation is known as
quantization noise.
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To minimize the quantization noise more bits are needed (i.e.,
more quantization levels)
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A trade off between BW and quality.
Voice Digitization
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Encoding: It is the fourth and final step in PCM.
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This is performed by a codec (coder/decoder).
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Three types of codecs exist:
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Waveform codecs
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Source codecs (vocoders)
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Hybrid codecs.
Voice Digitization
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Waveform codecs: It samples and codes the incoming
analog signal without regarding to how the signal was
generated.
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Quantized values of the samples are then transmitted to the
destination where the original signal is reconstructed.
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It is known for its simplicity with high quality output.
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Its disadvantage is that they consume considerably more BW
than other codecs.
Voice Digitization
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Source codecs (vocoders): match an incoming
signal to a mathematical model of how speech is
produced.
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Hybrid codecs: they are deployed in an attempt to
derive the benefits from both technologies. It
provide better voice quality at low BW than wave
codecs.
Popular Speech Codecs
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Codecs are best known for the sophisticated
compression algorithms they introduce into a
conversation.
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BW costs service providers money.
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The challenge for many SPs how to allow multiple
conversations to be carried on one 64kbps channel.
Popular Speech Codecs
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Table 2-1
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Table 2-2
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Page 19
Popular Speech Codecs
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G.711: is the best-known coding technique in use today.
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It is a wave form codec and is the coding technique used
in circuit-switched telephone networks.
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It has sampling rate of 8000 samples/sec.
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It requires a 12 bit per sample if uniform quantization is
used.
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It requires a 8 bit per sample if non-uniform quantization
is used.
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Both A-law and µ-law nonuniform quantization are used
with MOS of 4.3.
Disadvantage of G.711, it consumes 64Kbps in BW.
Popular Speech Codecs
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G.728 LD-CELP Code-Excited Linear Predictor (LD-CELP): codecs implement a
filter and contain a codebook of acoustic vectors.
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Each vector contains a set of elements in which the elements represent various
characteristics of the excitation signal.
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CELP coders transmit to the receiving end a set of information determining filter
coefficients, gain and a pointer to the chosen excitation vector.
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The receiving end contains the same code book and filter capabilities so that it
reconstructs the original signal.
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G728 is a backward –adaptive coder as it uses previous and current samples to
determine the applicable filter coefficients.
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G728 operates on 5 samples at one time. 5 samples at 8000 sample/sec are
needed to determine a code book vector and filter coefficients based on
previous and current samples.
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Given the coder operating on 5 samples at a time, a delay of less than 1 msec is
the result. Low delay means better voice quality.
Popular Speech Codecs
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G.723.1 ACELP: It can operate at either 6.3 Kbps (higher voice quality) or 5.3
Kbps.
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Bit rates are contained in the coder and decoder, and the transition between the
two can be made during conversation.
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The coder takes a bank-limited input speech signal speech signal that is
sampled a 8000 sample/sec and undergoes uniform PCM quantization, resulting
in a 16 bit PCM signal.
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The encoder works on blocks or frames of 240 samples at a time.
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Each frame corresponds to 30 milliseconds.
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With a look-ahead delay of 7.5 msec, the total algorithmic delay 37.5msec.
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G.723.1 gives an MOS of 3.8, which is highly advantageous in regards to the
BW used.
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The delay 37.5msec one way does present impediment to good voice quality,
but the final delay is depending on other network aspects and not only on the
codec used.
Popular Speech Codecs
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G.729: It operates at 8 Kbps. This coder uses frames of 10 msec,
corresponding to 80 samples at a sampling rate of 8000 sample/sec.
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This coder includes a 5-milisecond look-ahead. Resulting in an
algorithmic delay of 15 msec (better than G723.1).
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G729 uses an 80-bit frame. The transmitted bit rate is 8Kbps. With MOS
of 4.0.
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G729 is perhaps the best trade-off in BW for voice quality.
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A 64Kbps G711 codec is the standard in use on the PSTN.
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The other described codecs applied to VoIP as well
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VoIP engineers seeking to squeeze more conversations over valuable
BW by using codecs in compressing VoIP conversation over IP circuit.
Signalling
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Signaling describes the process of how calls are set up and torn down.
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Three main functions of signaling:
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Supervision: refers to monitoring the status of a circuit to determine if
there is traffic on the line.
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Alerting: deals with the ringing of a phone indicating the arrival of an
incoming call.
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Addressing: is the routing of a call over a network.
Signalling schemes
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Access signaling
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Core signaling
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B-ISDN signaling
Access Signalling
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Special requirements
Analogue Vs Digital
Telephones are simple terminals
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DTMF signaling
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Access Signalling
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ISDN Signaling
Access Signalling: Basic call establishment DSS1
Inter-exchange signalling: Signalling System 7 (SS7)
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Requirements for Inter-exchange
signaling different
Allocation of resources for connection
Role of Signaling System #7
Signalling System 7 (SS7)
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Most of Circuit switched networks, uses an in-channel signaling
where signaling follows the same path as the conversation.
This is called Channel-Associated Signaling (CAS).
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R1 Multifrequency (MF) used in N. America
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R2 Multifrequency Compelled (RFC) used in else where in the world.
Signalling System 7 (SS7)
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A newer signaling technology is out-of-channel signaling or called Common Channel
Signaling (CCS).
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CCS uses a separate transmission path for call signaling. This separation enables
the signaling to be handled in different manner to the call and also enables the
signaling to be managed by a network independent of the transport network.
Signalling System 7 (SS7): Examples of CCS systems
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Trunk networks
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CCS6 (old)
CCS7 (new)
Private networks
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DPNSS (Digital private Network Signaling Scheme) (old)
QSIG (new)
Signalling System 7 (SS7):Protocol Architecture
Signalling System 7 (SS7): Level 2 message types
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Link Status Signal Unit (LSSU): Indicates the status
of the link on which it is carries. Level 2 delivers
status of failed links to level 3
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Message Signal Unit (MSU): provides the structure
for transmitting higher layer information
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Fill in Signal Unit (FISU): acts as a flag in the SS7
network. Aids in synchronisation process
Signalling System 7 (SS7): Message Transfer Part level 3
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Level 3 provides three routing functions
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routing
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message discrimination
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distribution
Point code identifies unique nodes
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example 2001H, 2-68-4
Level 3 provides three management functions
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link management
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route management
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traffic management
Signalling System 7 (SS7): Point code
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Every point on the network has its own point address
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Address = point code
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Point codes only unique within a network and not globally
ITU provides 2 ways to define point codes
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basic: an number between 0 and 16,383
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International: 3 numbers: <zone:0-7><area:0-255><point:0-7>
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14 bits long (3-8-3 bits)
ANSI point codes:
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<network id.><cluster id.><network cluster member>
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24 bits long (8-8-8 bits)
Signalling System 7 (SS7): Standards
•There are a number of standards bodies
Signalling System 7 (SS7)
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Signaling System 7 (SS7) is the standard for CCS with
many national variants throughout the world,
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It routes control messages through the network to
perform call management (setup, maintenance and
termination) and network management functions.
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Although the network being controlled is ciecuit
switched, the control signaling is implemented using
packet switched network.
Signalling System 7 (SS7)
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The SS7 network and protocol are used for the following:
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Basic call setup, management, and tear down.
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Wireless services such as personal communications services
(PCS), wireless roaming, and mobile subscriber authentication.
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Local number portability (LNP)
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Toll-free (800/888) and toll (900) wireline services.
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Enhanced call features such as call forwarding, calling party
name/number display, and three-way calling.
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Efficient and secure worldwide telecommunications.
Signalling System 7 (SS7)
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Signaling Links SS7 messages are exchanged between network elements
over 56 or 64 Kbps bidirectional channels called signaling links.
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Signaling occurs out of band on dedicated channels.
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Out of band signaling provides:
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Faster call set up times,
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more efficient use of voice circuits,
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Support for Intelligent Network (IN) services that requires signaling to
network elements without voice trunks (such as database systems),
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Improved control over fraudulent network usage.
Signalling System 7 (SS7)
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Signaling Points Each signaling point in the SS7 network is uniquely
identified by a numeric point code.
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Point codes are carried in signaling messages exchanged between signaling
points to identify the source and destination of each message.
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Each signaling point uses a routing table to select the appropriate signaling
path for each message.
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Three kinds of signaling points are used in the SS7 network:
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Service switching points (SSP)
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Signal transfer points (STP)
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Service control points (SCP).
Signalling System 7 (SS7)
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Service Switching Points SSPs: are switches that originate,
terminate or tandem calls.
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An SSP sends signaling messages to other SSPs to setup, manage
and release voice circuits required to complete a call.
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An SSP may also send a query message to centralized database (an
SCP) to determine how to route a call (such as a toll-free 1-800/888
call in N. America).
Signalling System 7 (SS7)
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Service Control Points SCP:
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An SCP sends a response to the originating SSP containing the
routing number(s) associated with the dialed number.
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An alternating routing number may be used by SSP if the primary
number is busy or the call is unanswered within a specified time.
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Actual call features vary from network to network and from service to
service.
Signalling System 7 (SS7)
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Service Transfer Points STP:
Network traffic between signaling points may be routed via a packet
switched called an STP.
An STP routes each incoming message to an outgoing signaling link
based on routing information contained in the SS7 message.
Because STP acts as a network hub, an STP provides improved
utilization of the SS7 network by eliminating the need for direct links
between signaling points.
An STP may perform global title translation, a procedure by which the
destination signaling point is determined from digits present in the
signaling message (such as the dialed 800 number, the calling card
number, or mobile subscriber identification number).
An STP can also act as a firewall to screen SS7 messages
exchanged with other networks.
Signalling System 7 (SS7)
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Because the SS7 network is critical to call processing, SCPs and
STPs are usually deployed in mated-pair configurations in separate
physical locations to ensure network-wide service in the event of an
isolated failure.
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Links between signaling points are also provisioned in pairs. If one of
the links fails, the signaling traffic is re-routed over another link in the
linkset.
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Traffic is shared across all links in the linkset.
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The SS7 protocol provides both error correction and retransmission
capabilities to enable continued service in the event of signalling
point or link failures.
Signalling System 7 (SS7)
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SS7 Signaling Link Types: Signaling links are logically organized by
link type (A through F) according to their use in the SS7 signaling
network.
Signalling System 7 (SS7): SS7 Signaling Link Types
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Fig 2-6 & Table 2-3:
Signalling System 7 (SS7):Protocol Architecture
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A-LINK: Switch or Data Base Access to STP
B-LINK: Interconnects STP Pairs in different Regions
C-LINK: Connects Mated STP Pairs
D-LINK: SS7 Network Interconnect
E-LINK: Connects Switch to STP in another Region
F-LINK: Associated Signalling (direct switch to switch)
Signalling System 7 (SS7): SS7 Signaling Link
Types
A link
An A (access) link connects a signaling end point (an
SCP or SSP) to an STP. Only messages originating from
or destined to the signaling end point are transmitted on
an A link.
B link
B (bridge) links connect an STP to another STP. Typically,
a quad of B links interconnect peer (or primary) STPs (the
STPs from one network to the STPs of another). The
distinction between a B and D link is rather arbitrary. For
this reason, such links may referred to as B/D links.
C link
C (cross) links connect STPs performing identical
functions into a mated pair. They are used only when an
STPs has no other route available to a destination
signaling
Broadband signalling:
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Broadband signalling developed for
networks that based on ATM
Based on existing protocols
Extending signalling into private networks
Convergent Networks: Communication Services
ITU-T Multimedia Services: Framework
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Media component provides functions for user information
handling, such as information capture, presentation,
storage, transfer and post-processing.
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A communication task handles a set of media
components in a synchronized way to convey and
control complex information types
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Service level uses a set of communication tasks to
provide the communications services required by the
application level using the service platform
Service Platform
ITU-T F.700
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Media component provides functions for user information
handling, such as information capture, presentation,
storage, transfer and post-processing.
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A communication task handles a set of media
components in a synchronized way to convey and
control complex information types
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Service level uses a set of communication tasks to
provide the communications services required by the
application level using the service platform
Service Platform
ITU-T F.700
Generic Communication tasks
ITU-T F.700
Communication tasks
ITU-T F.700
Media Components
ITU-T F.700
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♦Audio
♦Video
♦Text
♦Graphics
♦Still pictures
♦Data
Media Components: Audio quality levels
ITU-T F.700
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♦ A(-1) quality is minimum to allow detecting
presence of speaker
♦ A0 quality is minimum needed to understand
speech from various speakers (at least G.711 or
G.728 codec)
♦ A1 is 7 kHz speech quality (at least G.722 codec
at 48kbps)
♦ A2 is 15 kHz broadcast quality sound/speech (at
least J.41 codec)
♦ A3 is 20 kHz hi-fi or CD quality
Media Components: Video quality levels
ITU-T F.700
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♦ V(-1) quality is minimum to allow detecting movement
♦ V0 is minimum videophone quality sufficient to show the head
to identify the person and to recognize facial expressions
♦ V1 is basic videophone quality sufficient to show the head and
shoulder with lips movements, and with the limited movement of
those of a seated person in normal conversation
♦ V2 is basic videoconference quality to a group of three seated
persons simultaneously, and with the limited movements of those
in a normal discussion
♦ V3 is television broadcasting quality (ITU-R BT.601)
♦ V4 is high definition television quality
Media Components: Text quality levels
ITU-T F.700
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♦ T0 is minimum quality with basic alphabets and punctuation,
but no formatting or choice of font
♦ T0 bis is videotext quality with basic alphabets and punctuation,
basic graphic character set, but no formatting or choice of font
♦ T1 is usable text conversation quality: font support for ISO10646-1 language area Latin-1 plus target language area; < 1
corrupted, dropped or marked missing character per 100; delay <
2s
♦ T2 is good text conversation quality: font support for all ISO10646-1 characters; < 1 corrupted, dropped or marked missing
character per 500; delay < 1s
Media Components: Still picture standards
ITU-T F.700
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
♦Facsimile standards in the basic version
have only 1 bit (black and white) per pixel
before compression, but can be extended to
gray levels or to colors
♦General still picture coding scheme such as
T.81 (ISO-JPEG) and T.82 (ISO-JBIG) with
various possible sets of parameters or
profiles
Media Components: Data
ITU-T F.700

♦ Data are usually organized in files:


􀂾 Software files are used to store or download
software
􀂾 Data files are not associated with any medium,
but may support other media components for
storage and transmission purposes
Middleware service elements
ITU-T F.700
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♦ Security: for authentication/non-repudiation;
privacy; integrity
♦ Directory
♦ Reservation
♦ Call control: for call set-up; in-call modification;
quality negotiation
♦ Charging and billing
♦ Media selection
♦ Conference control: for conference management;
multipoint protocol
Middleware service elements
ITU-T F.700 (Cont’)
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♦ Retrieval control: for browsing, navigation
♦ Mail control: for sending and receiving mail
♦ Poll control
♦ Application control: for navigation in a document;
device control
♦ Processing: for selection; assembly; translation;
media conversion
♦ Storage
♦ Replication
♦ Intercommunication
Value Chain
Source: UMTS Forum
Different Quality of Services
e.g., as defined in 3G Wireless

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♦ Conversational
􀂾 Adaptive Multi-rate Speech;
􀂾 Video Telephony
􀂾 H.323 and SIP
♦ Streaming
􀂾 web broadcast
􀂾 video streaming on demand
♦ Interactive
􀂾 Location-based services
♦ Background
􀂾 E-mail
􀂾 SMS
􀂾 Download
Service network
Telecommunication Services
TS 22.004
Basic Telecommunication Services
3G TS 22.105

♦ A bearer service provides the capabilities of
transmission of signals between access points



􀂾 A bearer has defined network function capabilities such
as capacity, delay and bit error rate, etc.
􀂾 User equipment requests the bearer capability from the
network
♦ A Teleservice provides the complete capabilities,
including terminal equipment functions, for
communication between users according to
protocols established by agreement among network
operators
Basic Telecommunication Services
3G TS 22.105
Supplementary Services
3G TS 22.004

♦A supplementary Service (3G TS 22.004)
complements and personalizes the usage of
basic telecommunication services.


􀂾Consequently, it cannot be offered as a
standalone service and must be offered together
with or in association with a basic
telecommunication service.
􀂾The same supplementary service may be
common to a number of telecommunication
services.
Bearer Services
3G TS 22.004


♦Multicast: Source determines the sinks
♦Broadcast: Source does not determine the
sinks
Bearer Service
QoS Range

♦An application may specify to the network its
end-to-end QoS requirements.




Data rate is amount of data transferred between
the two access points in a given period of time
Maximum transfer delay between the two access
point
Delay variation has to be controlled to support real
time applications
BER
Bearer Service
End-to-end Quality of Service (QoS) Classes
Bearer Service
QoS Range
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♦An application may specify to the network its
end-to-end QoS requirements.
Data rate is amount of data transferred between
the two access points in a given period of time
Maximum transfer delay between the two access
point
Delay variation has to be controlled to support
real time applications
BER
Session Initialization Protocol
Service Examples
SIP supports following traditional telecommunication services:
 Hold: Call Hold; Music on Hold/Call Park; Consultation Hold
 Transfer: Unattended Transfer; Attended Transfer
 Call Forwarding: Unconditional; - Busy; - No Answer
 3-way Conference
 Single Line Extension
 Find-Me
 Call Management: Incoming Call Screening; Outgoing Call
Screening
 Call Pickup
Convergent Networks
Communication Services
GPRS Services
GPRS Services

Point-to-Point


Point-to-Multipoint


One sender to one receiver
One sender to multiple receivers
Supplementary and value added services
Pont-to-Point Services
PTP

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PTP: Send data packets from one sender to one
receiver. Data arrival is confirmed. Lost packets are
resent upon request
PTP Connectionless Network Services (CLNS)
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Send data packets that are independent of each other
Example is IP
PTP Connection-Oriented Services (CONS)


Send sequence of connected packets through a logical
connection between sender and receiver
Example is X.25 protocol
Pont-to-Multipoint Services
PTM

PTM: send from one sender to multiple
receivers:

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
PTM Multicast
PTM Group Call
IP Multicast
PTM-Multicast
PTM-M
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Send to all users in a defined area (PTM-M)
Receivers do not need to be GPRS attached
Paging is not required
No confirmation of receipt of data, no
guarantee for reliable delivery of data
Repetition may increase the probability of
receiver to receive the data
Not ciphered
PTM Group Call
PTM-G
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
Send to a group of selected users within a
specified area
PTM-G may be:



Two-way connection: receiver may send data
back to sender
N-way connection: like a conference call
Receivers need to be GPRS attached and
are therefore known to the network
Internet Protocol Multicast
IP-M

Uses IP to exchange data, so the receiver groups
are not restricted to be within the PLMN but may be
anywhere on the Internet
Supplementary Services
Examples

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Unconditional forwarding of calls
Forwarding of calls when participants are not
reachable
Closed user group
Advice of charge
Value-Added Services
Examples

Valued added services are implemented on
top of the standard network services and help
the operators to differentiate themselves, e.g.


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Access Internet and Intranet
Use of e-mail and FAX
News
Traffic and weather information update
Commercial messages
Quality of Service

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♦Priority classes: high; middle; low
♦Guaranteed transmission rate: maximum;
average
♦Reliability of data transmission
♦Maximum time of delay
Convergent Networks
Communication Services
GPRS Services
QoS: Reliability

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GPRS specifies 3 classes each defining the
maximum rate of the following errors:
♦Loss of packet data
♦Transmission of a duplicate
♦Wrong order of packets
♦Unrecognizable transmission error
QoS: Reliability

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♦ Class 1: data is error sensitive
♦ Class 2: either data is less error sensitive or
error correction mechanism is in place
♦ Class 3: either data is error tolerant or relies
on other error correction
QoS: Delay


♦ Average delay is for all the packets to arrive
♦ 95% delay is the maximum delay for 95% of
the packets to arrive
QoS: Delay

♦In the delay specification in GPRS, the delays
includes those arising from:



􀂾Channel assignment procedure
􀂾Transmission over both the air interface and the
GPRS network
♦It excludes delays in external networks
(Internet; Intranet) and those caused by
service providers