Baohieu Va Dieukhien Ketnoi

8/21/2019 Baohieu Va Dieukhien Ketnoi

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Bo hiuviukhinktni

Signall ing and Connection Control

Gingvin: Hong TrngMinhEmail: [email protected]

Aug 2012

Bi gingmn hc
mailto:[email protected]:[email protected]


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cngmn hc

1.Tng quan vbo hiuviukhinktni

1.1 Giithiuchung

1.2 Giiphp iukhinhthngvinthng

1.3 Nguyn tciukhinhthngvinthng

1.4 Kintrc v phn loibo hiu

2. Bo hiutrong mngcnh

2.1 Kintrc mnghit

2.2 Hthngbo hius7

2.3 Giao thcbo hiuH.323

2.4 Giao thciukhincngphngtinH.248/Megaco

2.5 Giao thckhitophin SIP


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cngmn hc

3. Bo hiutrong mngthng tin d i ng

3.1 Tngquan vbo hiutrong mngdi ngtbo

3.2 Cc thtcbo hiumngtruy nhp

3.3 Thtcxl bo hiutrong mngli

4. Bo hiutrong phn haphngtinIP

4.1 Kintrc phn haphngtinIP

4.2 Bo hiukhitophin SIP

4.3 Thitlpphin trong IMS qua bo hiuSIP4.4 Cc giao thcbo hiukhc trong IMS


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cngmn hc

5. Bo hiuviukhinktnilin mng5.1 Cc giiphp cng nghtin tin

5.2 Cc chunktniv iukhintruy nhp

5.3 Cc giao thcbo hiulin mng

[1] Bi gingbo hiuv iukhinktni (angbin son)[2] Popovskij, Vladimir, Barkalov, Alexander, Titarenko, Larysa, Control and Adaptation inTelecommunication Systems, Springer, 2011.[3] John G. van Bosse, Fabrizio U. Devetak, Signaling in Telecommunication Networks,

second edition, John Wiley & Sons, Inc., 2007.[4] Travis Russell, THE IP MULTIMEDIA SUBSYSTEM (IMS): Session Control and OtherNetwork Operations,The McGraw-Hill, 2008.[5] Ralf Kreher, Torsten Ruedebusch, UMTS Signaling: UMTS Interfaces, Protocols,Message Flows and Procedures Analyzed and Explained, John Wiley & Sons, Inc., 2012.

Ti liutham kho


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1.Tng quan vbo hiuv iukhinktni

NB wireline LEX

RSU

MSCBSCGSM

SSP

SSP

SCP

Serv

SCP

Serv

Circuit Backbone

NB wireline

BB wireline

SGSN

GGSN

GSM

NAS

AAA

DSLAM

BAS

ATM

Switch

Packet Backbone @


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Services

Resources

Resour

ces

Transfer functional area

Transport managementfunctions

Service managementfunctions

Service controlfunctions

Transport controlfunctions

Infrastructu ral , appl icat ion, midd leware

and b aseware services

N

GNs

ervice

NGNt

ransport


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7

Trunk Gateway

Access Gateway

Resident Gateway

Signalling Gateway

TDM

SS7/ISUP

E1(voice)

SIGTRAN

MGCP

MGCP

MGCPSIPH323

VoIP Internet domain

E1/R2MFC

POTS

MGCP Phone

MGCP

POTS

APP

server

H323GK

SIPserver

BICC/SIPT
http://images.google.com.vn/imgres?imgurl=http://www.oki.com/jp/MMC/CTI/JIS/product/img/ip_phone_mktip.jpg&imgrefurl=http://www.oki.com/jp/MMC/CTI/JIS/product/pop03.htm&h=141&w=200&sz=18&tbnid=Fn6HvAf17xgJ:&tbnh=69&tbnw=97&start=20&prev=/images?q=sip+phone&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://www.oki.com/jp/MMC/CTI/JIS/product/img/ip_phone_mktip.jpg&imgrefurl=http://www.oki.com/jp/MMC/CTI/JIS/product/pop03.htm&h=141&w=200&sz=18&tbnid=Fn6HvAf17xgJ:&tbnh=69&tbnw=97&start=20&prev=/images?q=sip+phone&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://www.b2net.co.uk/i/sun_fire_12k.jpg&imgrefurl=http://www.b2net.co.uk/sun/sun_fire_12k_server.htm&h=250&w=153&sz=9&tbnid=xdg5b6bcL3oJ:&tbnh=104&tbnw=64&start=58&prev=/images?q=server+SUN&start=40&hl=vi&lr=&ie=UTF-8&sa=Nhttp://images.google.com.vn/imgres?imgurl=http://www.b2net.co.uk/i/sun_fire_12k.jpg&imgrefurl=http://www.b2net.co.uk/sun/sun_fire_12k_server.htm&h=250&w=153&sz=9&tbnid=xdg5b6bcL3oJ:&tbnh=104&tbnw=64&start=58&prev=/images?q=server+SUN&start=40&hl=vi&lr=&ie=UTF-8&sa=Nhttp://images.google.com.vn/imgres?imgurl=http://www.jumbonet.com/jumbocom/jumbo/multimed/phones/vidphone.jpg&imgrefurl=http://www-personal.umich.edu/~huangwei/VMC_research.htm&h=509&w=628&sz=32&tbnid=WTHBdO1zrMcJ:&tbnh=108&tbnw=133&start=4&prev=/images?q=video+phone&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://img.shopping.com/cctool/PrdImg/images/pr/177X150/00/01/4a/44/7a/21644410.JPG&imgrefurl=http://www.dealtime.co.uk/xPC-Cisco_CP_7960G_21644410&h=150&w=177&sz=5&tbnid=ninJMU-dnw0J:&tbnh=80&tbnw=94&start=34&prev=/images?q=sip+phone&start=20&hl=vi&lr=&ie=UTF-8&sa=Nhttp://images.google.com.vn/imgres?imgurl=http://www.oki.com/jp/MMC/CTI/JIS/product/img/ip_phone_mktip.jpg&imgrefurl=http://www.oki.com/jp/MMC/CTI/JIS/product/pop03.htm&h=141&w=200&sz=18&tbnid=Fn6HvAf17xgJ:&tbnh=69&tbnw=97&start=20&prev=/images?q=sip+phone&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://chat.audioadvisor.com/help/Generate/InternetCall(JavaClientOnly)withNetMeeting2.jpg&imgrefurl=http://chat.audioadvisor.com/help/Internet_Call_with_NetMeeti.htm&h=359&w=193&sz=17&tbnid=l7Lb1-b1Z5QJ:&tbnh=115&tbnw=62&start=13&prev=/images?q=netmeeting+client&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://nukefind.com/find/xlite.jpg&imgrefurl=http://nukefind.com/nuketalk.html&h=266&w=200&sz=12&tbnid=A7y14rvbVE8J:&tbnh=107&tbnw=81&start=19&prev=/images?q=sip+phone&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://www.oki.com/jp/MMC/CTI/JIS/product/img/ip_phone_mktip.jpg&imgrefurl=http://www.oki.com/jp/MMC/CTI/JIS/product/pop03.htm&h=141&w=200&sz=18&tbnid=Fn6HvAf17xgJ:&tbnh=69&tbnw=97&start=20&prev=/images?q=sip+phone&hl=vi&lr=&ie=UTF-8&sa=Ghttp://images.google.com.vn/imgres?imgurl=http://ascii24.com/news/i/hard/article/2003/03/14/thumbnail/thumb220x302-images711337.jpg&imgrefurl=http://ascii24.com/news/i/hard/article/2003/03/14/642478-000.html?geta&h=302&w=220&sz=5&tbnid=U_gbWCeHZ8wJ:&tbnh=111&tbnw=81&start=26&prev=/images?q=SIP+VoIP&start=20&hl=vi&lr=&ie=UTF-8&sa=Nhttp://images.google.com.vn/imgres?imgurl=http://www.b2net.co.uk/i/sun_fire_12k.jpg&imgrefurl=http://www.b2net.co.uk/sun/sun_fire_12k_server.htm&h=250&w=153&sz=9&tbnid=xdg5b6bcL3oJ:&tbnh=104&tbnw=64&start=58&prev=/images?q=server+SUN&start=40&hl=vi&lr=&ie=UTF-8&sa=Nhttp://images.google.com.vn/imgres?imgurl=http://www.dataweek.co.za/articles/Dataweek%20-%20Published%20by%20Technews/w6351.jpg&imgrefurl=http://www.dataweek.co.za/news.asp?pklNewsID=12146&pklIssueID=352&pklCategoryID=37&h=100&w=185&sz=5&tbnid=C85exuq-1wUJ:&tbnh=51&tbnw=94&start=36&prev=/images?q=server+ADVANTECH&start=20&hl=vi&lr=&ie=UTF-8&sa=N


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Anyt ime, Anywhere, Any service

Broadband Internetat home

EnterpriseIntranet

Public

Wireless LANHotspots

Public

GSM/GPRSUMTS

Mobile PDA

PCPSTN/ISDN

at home


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Signaling is used between user and the network, orbetween two network elements to exchange

various control information like:

+ Traffic Descriptors+ Service Descriptors

+ Channel Identifiers

In other words, Signaling is used to dynamically

establish, monitor, and release Connections(including physical, virtual and logical connections).


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Signaling is used only for establishment andrelease of dynamic connections

Static connections are configured, manually or

otherwise, and may or may not require signaling.

Signaling provides the means for resource

reservation.

In essence, Signaling provides the means to

exchange connection-related information prior toand/or after information transfer.


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Traditionally, use of signaling in Telecom Networkswas bare minimum.

It was restricted to establish/release a voice

channel in order to allow telephonic conversation.

Now, with advent of supplementary services (e.g.,

CLIP, Call Forwarding, etc.), signaling is becoming

more complex.

For e.g., SS7 which has an advanced networkarchitecture provides feature-rich signaling.


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If Permanent Virtual Circuits (PVCs) areestablished, generally, no signaling is required.

For Switched Virtual Circuits (SVCs), signaling

takes place using well-defined signaling protocol.

The signaling complexity is dependent upon theunderlying technology.

For e.g., Q.2931/Q.2971 (signaling protocol for

ATM) is much more complicated vs Q.933

(signaling protocol for frame relay).


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Datagram networks, generally, do not requiresignaling. This is because by very definition, a

connectionless network does not entail

connection setup.

To provide QoS, some of resource reservation

and hence some form of signaling is required.

For e.g., newer protocols like MPLS and RSVP

require some form of signaling messageexchange and resource reservation.


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Telecom Network:In band signaling refers to using the same voice

frequency band to carry signaling information as

that used to carry voice (i.e., 300-3400Hz).

In contrast, out band signaling refers to using

frequencies above the voice band (but below the

upper threshold of 4000Hz) to carry signaling

information.


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Datacom Network:In band signaling refers to using the same

virtual channel to carry signaling information

as that used to carry data.

In contrast, in Out band Signaling the

signaling information and data are carried on

different virtual channels.


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Telecom Network:

In Inchannel signaling, the same physical channel

carries signaling information as well as voice and

data.In contrast, Common Channel Signaling uses a

separate channel for solely carrying signaling

information for a number of connections.


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Datacom Network:

To some extent, inchannel signaling and common

channel signaling in telecommunication networks

is analogous to inband signaling and outbandsignaling of data communication networks

respectively.


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Telecom Network:

Both these techniques are variants of Common

Channel Signaling

In Associated signaling, the signaling channelsand the data paths pass through the same

network elements.

In Non-associated signaling, there is no

correspondence between signaling channelsand data paths.


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Telecom Network:

Both these techniques are variants of Common

Channel Signaling

In Associated signaling, the signaling channelsand the data paths pass through the same

network elements.

In Non-associated signaling, there is no

correspondence between signaling channelsand data paths.


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Datacom Network (for ATM):

In Channel associated signaling, all the

signaling messages for each VP is exchanged

on VCI=5 of that virtual path.In Channel non-associated signaling, all the

signaling messages of all the virtual paths are

exchanged on VPI=0 and VCI=5.

Associated Signaling versus Non-AssociatedSignaling


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Another technique metasignaling finds mention

in various signaling standards.

Metasignaling refers to the process of

establishing signaling channels usingsignaling procedures.

The signaling channel so established is then

used to establish channels for data transfer.


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Acknowledgements

Timer protection

Parameter negotiation

Call/Connection identification

Finite state machine modelling

Message encoding and decoding (TLV format)


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Required due to unreliable nature of transmission

media.

The classical two-army problem suggests that no

scheme can provide full-proof acknowledgement

for an unreliable media. However, 2 or 3

handshakes is typically sufficient for a normal case.


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Timers are used to avoid inordinate delays in case

the signaling messages get lost or corrupted.

Timer is started after message transmission.

In case message is lost or discarded, the timer

expires and message is retransmitted.


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If the message reaches safely and is acknowledged,the timer is stopped.

Choosing the correct timeout value is important.

If this value is too small, then timers will timeout very

frequently.If a very large value is chosen, it may defeat the

purpose of keeping timers.

Typical value is twice the round-trip propagation

time..


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This entails arriving at a common set of parameters.

The nature and scope of parameter negotiation

depends on the number of handshakes.

In a two way handshake, negotiation is bare

minimal.

A three way handshake provides more scope for

negotiation.


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Consider the following steps:An end-system A sends a connection establishment

request to B

A sends another request to end-system B for

connection establishmentA then receives a reply from B. How does A identify

to which request has B replied to?

The solution is to generate a uniq_num and

accompany it with every message


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A Call goes through three phasesCall establishment

Data transfer

Call releasing

The FSM accepts messages only if the message ispermitted in that state.

The state change happens when 1) a message is

received from peer, 2)Timer expires and 3) User

Request. received from User.


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Generally, messages are encoded in Type-Length-

Value (TLV) format.

Type: Identifies the type

Length: Length of message (total length or length

excluding the header)

Value: The actual contents

Information blocks within the message may also be

encoded in TLV format.


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Point-to-Point signaling model is used to establish

and release connections between two end-points.

The is the most common model of signaling.

Not only is this model popular, it is also very simple

to implement.


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Point-to-Multi Point signaling model is used toestablish and release connections between a root

and multiple end-points.

This form of signaling is mainly used for multicasting

or broadcasting applications (e.g., distant learning).


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A PMP call is generally started by the root.The root may take this step voluntarily,

Or, it may do the same after receiving an explicit

request from a leaf.

The leaf can send the connection establishment

request to the root through signaling channel or

through other means.

The first connection is established following point-to-point procedures


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After a PMP call is established, parties are addedby the root.

The root is informed either through a signaling

message, or through some other means.

Subsequent parties have no say in determining

the parameters of the connection, as it has

already been fixed


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A leaf of a point-to-multipoint call can drop itselfout of the connection by sending a message to

the root.

It is mandatory for the root to entertain this

request, and drop that particular party.

If the root drops itself out of the connection, the

whole connection is cleared.


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By definition, a PMP call is one in which the dataflows from the root to the leaves (i.e.

unidirectional in nature).

Theoretically, nothing precludes bi-directional

data-flows in PMP calls.

However, if leaves are allowed to send data to the

root, there is a multipoint-to-point connection

along with the point-to-multipoint call.


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Advantages+ Saving of network resources like bandwidth. The

% saving depends upon the breadth and the depth

of the PMP tree.

Disadvantages

- PMP connections are difficult to establish,

manage, and release.

- It is not easy for the leaves to indicate the root tostart a connection.

- The unidirectional nature


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2. Giiphp iukhinhthngvinthng

Introduction into Control Technologies for QoS Providing

telecommunication systems (TCS)

the practical QoS includes the following characteristics: delay of connection;

data throughput; transmission quality.

Logic structure for formation of SLA


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Introduction into Control Technologies for QoS ProvidingTechniq ues for the con trol plane.

The technique CAC (Call Admission

Control) controls new requests for traffic

transmission through the network.

The technique of QoS routing provides the

choice of a route satisfied to a required

quality of service for a given data flow

The third technique is called the resource

reservation.


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Introduction into Control Technologies for QoS ProvidingTechniq ues fo r data plane.

The technique of buffer management is

reduced to the control of packages which

are waiting in a transmission queue. RED

The mechanism of congestion avoidance

support the level of networks loading

a bit below of its throughput.

The approach of packet marking is used to

denote a particular service level for

different packages.


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The main goal of techniques from the group of queuing and scheduling is the

choice of a package for transmission from the buffer into a communication channel.

The majority of service procedures (disciplines, or schedulers) is

based on the rule FIFO (first in first out).

discipline of privileged (high-priory) service, example is a mechanism of

weighted fair queuing (WFQ)

discipline which is based on classification of data flows according withtheir classes of services. It is named CBQ (Class-Based Queuing) .


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The techniques of the traffic classification are connected with classification of

packages on the input of a network.

The goal of traffic shaping is control of speed and size of data flows from the

input of a network. a leaky bucket and a token bucket

The management p lane of QoS includ es techniq ues respo nsib le for

maintenance, admin istrat ion, and con trol of a network in respect to usertraff ic.


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Introduction into Control Technologies for QoS Providing

Basic components of models for service providing


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Peculiarities of RACS ApproachThe Resource and Admission Control Sub-System (RACS) executes the control

functions for access network and boundary node of the kernel execution level.

The component A-RACF (Access Re-source and Admission Control Function)

executes some functions connected with the access to network resources, as wellas control functions of these resources.

The component SPDF (Service-Based Policy Decision Function) executes

control using the access politics to service on the boundary of the level of the

network kernel.


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Peculiarities of RACS Approach

Architecture of RACS subsystem

RCEF.

Resource Control

Enforcement

Function.

C-BGF Core

Border

Gateway

Function

Network Attachment Subsystem (NASS)


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The service level takes responsibility for the exchange of

signaling messages among the applications.

The transport level is responsible for both reliable transmissionof the data packages and trafficscontrol.

The ITU determines the management architecture of QoS in its

standard. The main idea of the management architecture of QoS is

independence of the transport level from the service level.

the speech is transmitted using the IP protocol through the Internet

The traffic can run further through a network of a mobile operator.


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Architecture of RACF subsystem

The function of RACF determines the availability of

network resources and executes their control.


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Subscriber signalling

PSTN

Switching in

exchanges


(analog or ISDN=DSS1)

Network-

internalsignalling

(SS7)

Transmission

(PDH, SDH)

Databases in

the network

(HLR)


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Analog subscriber signalling

The calling party (user A) tells the local exchange to set up

(disconnect) a call by generating a short (open) circuit in the

terminal => off-hook (on-hook) operation.

The dialled called party (user B) number is sent to the local

exchange in form of Dual Tone Multi-Frequency (DTMF) signalbursts.

Alerting (ringing) means that the local exchange sends a strong

sinusoid to the terminal of user B.

In-channel information in form of audio signals (dial tone, ringbacktone, busy tone) is sent from local exchange to user. User can

send DTMF information to network.

1

2

3

4


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Analog subscriber signalling in action

LE AUser A User B

Ringing

signal

Off-hook

(user B

answers)

Off-hook SS7

signalling

(ISUP)Dial tone

B number

Ringback

tone (or busy

tone)

LE B

Connection established

LE = local exchange


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ISDN subscriber signalling in action

LE AUser A User B

Ringing

Off-hook

(user B

answers)

Off-hook SS7

signalling

(ISUP)B number

Tones

generated in

terminal

LE B

Setup

Call proc Setup

Alert

Conn

Alert

Conn

DSS1 signalling

messages

Connection established


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What does ISDN originally mean?

1. End-to-end digital connectivity

2. Enhanced subscriber signaling

3. A wide variety of new services (due to 1 and 2)

4. Standardized access interfaces and terminals

ISDN is not a new network separated from the PSTN. Interworkingwith normal PSTN equipment is very important.

ISDN

terminal

PSTN

terminal

interaction is

possible

Idea originated inthe 1980s


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PSTN vs. ISDN user access

300 3400 Hz analog transmission band

Poor-performance subscriber signaling

2 x 64 kbit/s digital channels (B channels)

16 kbit/s channel for signaling (D channel) => Digital

Subscriber Signalling system nr. 1 (DSS1)

PSTN

Basic

Rate

AccessISDN

Primary

Rate

AccessISDN

30 x 64 kbit/s digital channels (B channels)

64 kbit/s channel for signaling (D channel)

Mainly used for connecting private branch exchanges(PBX) to the PSTN.


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End-to-end digital signalling

ISUPQ.931

Q.921

I.430

Q.931

Q.921

I.430

MTP 3

MTP 2

MTP 1

Q.931

Q.921

I.430

Q.931

Q.921

I.430

ISUP

MTP 3

MTP 2

MTP 1

contains the signalling messages for call control

User interface User interfacePSTN Network

DSS1

SS7

DSS1


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Signalling System nr. 7 (SS7)

PSTN

Switching in

exchanges


(analog or ISDN=DSS1)

Network-

internalsignalling

(SS7)

Transmission

(PDH, SDH)

Databases in

the network

(HLR)


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History of inter-exchange signalling

SS6 = CCIS (common channel interoffice signaling) was deployed

in North America as an interim solution, but not in Europe. CCISis not the same thing as SS7.

Starting from 1980 (mainly in Europe), CAS was being replaced

by SS7. The use of stored program control (SPC) exchanges

made this possible. Like CCIS, signalling messages are

transmitted over separate signalling channels. Unlike CCIS, SS7technology is not monolithic, but based on protocol stacks.

Before 1970, only channel-associated signalling (CAS) wasused. In CAS systems, the signalling is carried in-band along

with the user traffic.

CAS

CCIS

SS7


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Channel-associated signalling (CAS)

CAS means in-bandsignalling over the same physical channels as the

circuit-switched user traffic (e.g. voice).

Signalling to/from databasesis not feasible in practice(setting up a

circuit switched connection to the database and then releasing it would

be extremely inconvenient).

Exchange Exchange

Exchange

Circuit switched connection

Signalling is possible

Signalling is not possible beforeprevious circuit-switched link is

established

CAS has two serious draw-backs:

Setting up a circuit switched connection is very slow.


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Common channel signalling (CCS)

In practice, CCS = SS7.

Exchange Exchange

Signalling is possible anywhere anytimeDatabase

End-to-end signallingis possiblebefore call setup and also during the

conversation phase of a call.

The packet-switched signalling network is totally separated from the circuit-

switched connections. Consequently:

Signallingto/from databases is possibleanytime.

There is one drawback: It is difficult to check if the circuit-switched

connections are really working (= continuity check).


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Signalling example

ExchExchUser A

(calling

user)

Database

A typical scenario:

User A calls mobile user B. The call is routed to a specific gateway

exchange (GMSC) that must contact a database (HLR) to find outunder which exchange (MSC) the mobile user is located. The call is

then routed to this exchange.

OuluTokyo

London

User B

(called

user)

Exch


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ISDN User Part

(ISUP)

Protocol layers (levels) of SS7

MTP level 3 (routing in the signalling network)

MTP level 2 (link-layer protocol)

MTP level 1 (64 kbit/s PCM time slot)

Signalling Connection

Control Part (SCCP)

Transaction Capabilities Application

Part (TCAP)

Application protocols (e.g. Mobile

Application Part, MAP)

MTP

MTPuser

SS7 application protocol

for managing circuit-switched connections


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MAP

ISUP

TCAP

SCCP

SS7 protocols vs. OSI model

MTP level 3

MTP level 2

MTP level 1

Application

Presentation

Session

Transport

Network

Data link

Physical

SS7 protocol stack OSI protocol layermodel


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OSI protocol layer model

Multiplexing & transport of bits, time slots

inPDH or SDH systems

Switching & routingthrough the

communications network

Link-layer flow & error control

End-to-end flow & error control

User application(in this case, the

actual signalling messages)

Data compression & coding

Dialogue control

Application layer

Presentation layer

Session layer

Transport layer

Network layer

Data link layer

Physical layer


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Message Trasfer Part (MTP) functions

MTP level 1 (signalling data link level):Digital transmission channel (64 kbit/s TDM time slot)

Frame-based protocol for flow control, error control (using Automatic

Repeat reQuest, ARQ), and signalling network supervision andmaintenance functions.

Routing in the signalling network between signalling points (using

signalling point codes).

MTP level 3 users are ISUP and SCCP (other users such as TUPor DUP are not widely used any more).

MTP level 3 (signalling network level):

MTP level 2 (signalling link level):


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MTP level 2 frame formats

F CK SIF SIO LI Control F

F CK SF LI Control F

F CK LI Control F

MSU (Message Signal Unit)

LSSU (Link Status Signal Unit)

FISU (Fill-In Signal Unit)

Level 3 user information

Network:

NationalInternational

User part:

ISUPSCCPSignalling

networkmanagementMSBLSB


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MTP level 2 frames

MSU (Message Signal Unit):Contains actual SS7 signalling messages

The received frame is MSU if LI > 2

(LI = number of octets)

LSSU (Link Status Signal Unit):Contains signalling messages for MTP level 2 (signalling

link) supervision

The received frame is LSSU if LI = 1 or 2

FISU (Fill-In Signal Unit):

Can be used to monitor quality of signalling link at

receiving end

The received frame is FISU if LI = 0

S (S ) SS


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Signalling points (SP) in SS7

Network elements (relevant from signalling point of view) contain signallingpointsidentified by unique signalling point codes.

Exchange

STP

SP

SP

STP

Signalling Point (in a database, suchas HLR in mobile network)

Signalling Transfer Pointsonly relay signalling messages

Signalling Point(signalling

termination in an exchange)

STP

MAP

ISUP

Si lli i t d (SPC)


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Signalling point code (SPC)

SS7 signalling messages contain MTP level 3 routing information in the form

of a routing label:

SIO octet

DPC

DPC

LSBMSB

OPC

OPC

OPC SLS

Signalling message

payload

International (and most national) signalling

networks (ITU-T):

14-bit Destination Point Code (DPC)

14-bit Originating Point Code (OPC)

4-bit Signalling Link Selection (SLS)

North American national signalling

network (ANSI):

24-bit DPC and OPC, 5-bit SLS code

Format for international SPC:

Zone Area/Network SP

3 bits 3 bits8 bits

For examples, see:www.numberingplans.com

S SPC b d t diff t t k


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Same SPCs can be reused at different network

levels

SPC = 277

SPC = 277

International

National

SPC = 277 means different signalling points (network elements) at

different network levels.

F CK SIF SIO LI Control F

The Service Information Octet (SIO)indicates whether the DPC and OPCare internationalor nationalsignalling point codes.

ISDN U P t (ISUP)


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ISDN User Part (ISUP)

ISUP is a signalling application protocol that is used for establishing and

releasing circuit-switched connections (calls).

Only for signalling between exchanges (ISUP can never be used

between an exchange and a stand-alone database)

Not only for ISDN (=> ISUP is generally used in the PSTN)

Structure of ISUP message:

SIO (one octet)

Routing label (four octets)

CIC (two octets)

Message type (one octet)

Mandatory fixed part

Mandatory variable part

Optional part

Must always be included in ISUP message

E.g., IAM message

E.g., contains called (user B) number in IAM

message

ISUP i lli


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ISUP signalling messages

Basic ISUP signalling messages:

Call setup:

IAM (Initial address message)

ACM (Address complete message)

ANM (Answer message)

From LE A to LE B

From LE B to LE A

Call release:

REL (Release message)

RLC (Release complete message)

Direction depends onreleasing party (user

A or user B)

Diff b t SLS d CIC


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Difference between SLS and CIC

The four-bitsignalling link selection (SLS) field in the routing label

defines the signalling linkwhich is used for transfer of the signallinginformation.

The 16-bitcircuit identification code (CIC) contained in the ISUP

messagedefines the TDM time slot or circuitwith which the ISUP

message is associated.

Exchange

STP

Exchange

Circuit

Signalling link

Si lli i IAM


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Signalling using IAM message

Exchange ExchangeExchange

SPC = 82

Circuit

14

SPC = 22 SPC = 60Circuit

20

STP

SL 4

SL 7

STP

Outgoing message:OPC = 82 CIC = 14

DPC = 22 SLS = 4

Processing in (transit) exchange(s):Received IAM message contains B-number. Exchange

performs number analysis(not part of ISUP) and selects

new DPC (60) and CIC (20).

S t f ll i ISUP


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Setup of a call using ISUP

LE A LE BTransit exchangeUser A User B

Setup IAMIAM

Setup

Alert

Connect

ACM

ANM

ACM

ANM

Alert

Connect

Charging of call starts now

Number analysisDSS1

signalling

assumed

Call set p Signalling seq ence 1


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Call setup: Signalling sequence 1

User A User B

Off hook

Dial tone

B number

Local exchange detects setup request

and returns dial tone

Local exchange:

analyzes B number

determines that call should

be routed via transitexchange (TE)

LE A LE BTE



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User A User BLE A LE BTE

Initial address message (IAM)

ISUP message IAM is sent to transit exchange (TE).

TE analyzes B number and determines that call should be

routed to local exchange of user B (LE B).

IAM message is sent to LE B.

There now exists a circuit-switched path (the path is cutthrough) between user A and LE B.



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Address complete

message (ACM)Ringing signalRingback

tone

Ringing signal is sent to user B (=> user B is alerted).

Ringback tone (or busy tone) is sent to user A.

(Ringback/busy tone is generated locally at LE A or is sent fromLE B through circuit switched path.)

or



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Answer message (ANM)User B answers

User B answers, connection is cut through at LE B.

Charging of the call starts when ISUP message ANM is received

at LE A (the normal case).

The 64 kbit/s bi-directional circuit switched connection is now

established.

Charging

starts now

Conversation over this pipe

E 164 numbering scheme


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E.164 numbering scheme

00

0

358 9

9

1234567

1234567

1234567

International number

National number

User number

Prefix

Country code

Area code

358

9

In each exchange, the B number is analyzed at call setup (afterthe IAM message containing the number has been received) and

a routing program (not part of ISUP) selects the next exchange to

which the call is routed.

or mobile network code, e.g. 40

E 164 number structure


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E.164 number structure

00 358 9 1234567

Prefix

For examples, see:www.numberingplans.com

Country code (1-

3 digits)

National destination code (1-3 digits)

Max. 15 digits

Subscriber number

Area code, e.g. 9

Mobile network code, e.g. 40

MSISDN number

Signalling sequence for call release


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Signalling sequence for call release


On hookRelease message (REL)

Release complete message (RLC)

The circuits between exchanges are released one by one.

(The generation of hanging circuits should be avoided, sincethese are blocked from further use.)

Charging

stops

Conversation over this pipe

Signalling Connection Control Part (SCCP)


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Signalling Connection Control Part (SCCP)

SCCP is required when signalling information is carried between exchanges

and databases in the network.

An important task of SCCP is global title translation (GTT):

STP DatabaseExchange

STP with GTT capability

Exchange knows the global title (e.g. 0800 number or IMSI number

in a mobile network) but does not know the DPC of the database

related to this global title.

1.

SCCP performs global title translation in the STP (0800 or IMSInumber => DPC) and the SCCP message can now be routed to the

database.

2.

Why GTT in STP network node?


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Why GTT in STP network node?

Global title translation (GTT) is usually done in an STP.

Advantage: Advanced routing functionality (= GTT) needed only in a

fewSTPs with large packet handling capacity, instead of many

exchanges.

Exchange

STP

Database

Exchange

Exchange

ExchangeExchange

Database

Exchange

Example: SCCP usage in mobile call


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Example: SCCP usage in mobile call

SCCPSCCP

MSC located in Espoo HLR located in Oslo

STP

SPC = 82 SPC = 99

SPC = 32

SCCP/GTT functionality

Outgoing message:

OPC = 82 DPC = 32

SCCP:IMSI global title

Processing in STP:

Received message is given to SCCP for GTT.

SCCP finds the DPC of the HLR: DPC = 99

Mobile switching center (MSC) needs to contact the home location register

(HLR) of a mobile user identified by his/her International Mobile Subscriber

Identity (IMSI) number.

To sum it up with an example


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To sum it up with an example

PSTN

Typical operation of a

local exchange


(analog or ISDN=DSS1)Network-

internalsignalling

(SS7)

Transmission

(PDH, SDH)

Databases in

the network

(HLR)

Part B, Section 3.3 in UnderstandingTelecommunications 2

Basic local exchange (LE) architecture


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Basic local exchange (LE) architecture

Time

switch

TDM links to

other

network

elements

Switch control

Switching system

E.164 number analysisCharging

User databases

LIC

LIC

Tone

Rx

Group

switch

Sign.

ETC

ETC

Exchange

terminal

circuit

Line

interface

circuit

SS7 Signallingequipment

Control systemO&M functions

Subscriber stage

Modern trend: Switching and control functions are separated into different

network elements (separation of user and control plane).

Tone generator

Setup of a call (1)


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Setup of a call (1)

Time

switch

2. Check user database. For instance, is

user A barred for outgoing calls?

Switching system

3. Reserve memory for user B number

LIC

LIC

Tone

Rx

Group

switch

Sign.

ETC

ETC

Control system

Phase 1. User A lifts handset and receives dial tone.

1. Off hook

Local exchange of user A

4. Tone Rx is connected

5. Dial tone is

sent

(indicating

network isalive)

Tone generator

Setup of a call (2)


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Time

switch

3. Number analysis

Switching system

4. IN triggering actions? Should an externaldatabase (e.g. SCP, HLR) be contacted?

LIC

LIC

Tone

Rx

Group

switch

Sign.

ETC

ETC

Control system

Phase 2. Exchange receives and analyzes user B number.

2. Number (DTMF

signal) received1. User A dials

user B number

Setup of a call (2)


Setup of a call (3)


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Time

switch

2. Outgoing circuit is reserved

Switching system

LIC

LIC

Tone

Rx

Group

switch

Sign.

ETC

ETC

Control system

3. Outgoing signalling message (ISUP IAM)contains user B number

Phase 3. Outgoing circuit is reserved. ISUP Initial address message (IAM) is

sent to next exchange.

Setup of a call (3)

1. Tone receiver

is disconnected


E.g.,

CIC = 24

IAM

(contains

information

CIC = 24)

Setup of a call (4)


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Time

switch

1. ISUP ACM message indicates free or busy

user B

Switching system

LIC

LICGroup

switch

Sign.

ETC

ETC

Control system

3. Charging starts when ISUP ANM message

is received

Phase 4. ACM received => ringback or busy tone generated. ANM received

=> charging starts.

Setup of a call (4)


ACM,

ANMTone generator2. Ringback or

busy tone is

locally

generated

4. Call

continues

H323 protocols and mechanisms


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95

H.323 protocols

H.235Security within H.245-based systems

H.245Interworking with the PSTN

H.450.xSupplementary services

H.460.xVarious H.323 protocol extensionsH.501Protocol for mobility management and inter/intra-domain

communication

H.510User, terminal, and service mobility

H.530Security specification for H.510

p



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96

H.323 protocols

H.323

IP

UDP

RTP

RTCP

TCP/UDP TCP UDPUDP TCP

Audio

Codecs

G.711G.723.1

G.729

..

Video

Codecs

H.261H.263

H.264

..V.150 T.120

TCP/UDP

T.38

H.225.0

Call

Signaling

H.245H.225.0

RAS

Terminal Control and ManagementData

ApplicationsMedia Control

Multimedia Applications, User Interface

Typical H.323 Stack

p



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97

H.323 protocols

PSN

CSN

V.70Terminal

H.324Terminal

SpeechTerminal

H.322Terminal

SpeechTerminal

H.320Terminal

H.321Terminal

GSTNGQOS

LANN-ISDN B-ISDN

H.323

MCU

H.323

Terminal

H.323

Gatekeeper

H.323

Gateway

H.323

Terminal

H.323

Terminal

H.323 Network Elements

Elements

p



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98

H.323 protocols

Endpoint

Gatekeeper

Endpoint

Gatekeeper

RASRAS

Signalling (Q.931)

H.245

RTP/RTCP

Q.931/H.245

Q.931/H.245

Q.931/H.245

Annex G

Gatekeeper Routed Signaling

Direct Routed Signaling

p



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99

H.323 protocols

Terminals

Multipoint Control Units (MCUs)

Gateways

Gatekeeper

Border Elements / Peer Elements

Referred to as

endpoints

Elements

p



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100

H.323 protocols

Elements

Telephones

Video phones

IVR devices

Voicemail Systems

Soft phones (e.g., NetMeeting)

Terminals

p



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101

H.323 protocols

Elements

Responsible for managing multipoint conferences (two

or more endpoints engaged in a conference)

The MCU contains a Multipoint Controller (MC) thatmanages the call signaling and may optionally have

Multipoint Processors (MPs) to handle media mixing,

switching, or other media processing

MCU



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102

H.323 protocols

Elements

The Gateway is composed of a Media Gateway Controller (MGC)

and a Media Gateway (MG), which may co-exist or exist

separately

The MGC handles call signaling and other non-media-related

functions

The MG handles the media

Gateways interface H.323 to other networks, including the PSTN,

H.320 systems, other H.323 networks (proxy), etc.

GW



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103

H.323 protocols

ElementsGatekeeper

The Gatekeeper is an optionalcomponent in the H.323

system which is used for admission control and address

resolution

The gatekeeper may allow calls to be placed directly

between endpoints or it may route the call signaling

through itself to perform functions such as follow-

me/find-me, forward on busy, etc.

GK



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104

H.323 protocols

RAS

GK GK

T T MCU GW

GK

GK GK

GWGWGWGW

GK

Hierarchical Gatekeeper

topology is widely used,but explicitly discussed in

H.323

Topology with RAS



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105

H.323 protocols

RAS

Registration, Admission, and Status

Used between the endpoint and its Gatekeeper in order to

Allow the Gatekeeper to manage the endpoint

Allow the endpoint to request admission for a call

Allow the Gatekeeper to provide address resolution functionality for

the endpoint

RAS signaling is required when a Gatekeeper is present in the

network (i.e., the use of a Gatekeeper is conditionally mandatory)



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106

H.323 protocols

RAS

RAS messages generally have three types

Request (xRQ)

Reject (xRJ)

Confirm (xCF)

Exceptions are

Information Request / Response / Ack / Nak

The nonStandardMessage

The unknownMessage response

Request in Progress (RIP)Resource Available Indicate / Confirm (RAI/RAC)

Service Control Indication / Response



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107

H.323 protocols

RAS

Typically, RAS communications is carried out via UDP

through port 1719 (unicast) and 1718 (multicast)

For backward compatibility sake, an endpoint should be

prepared to receive a unicast message on port 1718 or 1719Only UDP is defined for RAS communications

GRQ and LRQ may be send multicast, but are generally

sent unicast

All other RAS messages are sent unicast

RAS Port



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108

H.323 protocols

RAS

When an endpoint comes to life, it should try to discover a

gatekeeper by sending a GRQ message to a Gatekeeper

Address of a Gatekeeper may be provisioned

The endpoint may send a multicast GRQAddress of a Gatekeeper may be found through DNS queries (Annex

O/H.323)

There may be multiple Gatekeepers that could service an endpoint,

thus an endpoint should look through potentially several GCF/GRJ

messages for a reply

Gatekeeper Request - GRQ



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109

H.323 protocols

RAS

If a Gatekeeper does not wish to provide service to the

endpoint, it will generally send a GRJ message to the

endpoint

As a security consideration to avoid DoS attacks, one might

want to consider ignoring requests from unknown endpoints

The GRJ message will carry one of several rejection

reasons

Gatekeeper Reject - GRJ



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110

H.323 protocols

RAS Gatekeeper Confirm - GCF

If the Gatekeeper wishes to provide service to the

endpoint, it will return a GCF message

The GCF message will contain a number of dataelements that will later be used by the endpoint



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111

H.323 protocols

RAS

Address resolution Address resolution Propagation of routing

information

Usage reporting

Access authorization (useful forclearinghouses and non-

clearinghouse environments)

A Border Element or PeerElement can act as an

aggregation point (usually the

role of a Border Element)

RAS

AnnexG/H.2

25.0

RAS LRQ versus Annex G



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112

H.323 protocols

RAS

GWGW

Setup

Call Proceeding

Progress

Alerting

Connect

CONNECTED

Option

al

Release Complete

Basic Call Setup Signaling

Overview of Session Initiation Protocol


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SIP - Introduction

SIP architecture and philosophy

Overview of Working

Methods Used in SIP

SIP messages & responses

Security

Summary

References

Introduction


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SIP is

An Application-layer control (signaling) protocol forcreating, modifying and terminating sessions with oneor more participants.

Sessions include Internet multimedia conferences,Internet telephone calls and multimedia distribution.

Members in a session can communicate via multicast orvia a mesh of unicast relations, or a combination ofthese.

Text based , Model similar to HTTP : uses client-servermodel

SIP Basic Functionality


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Supports 5 facets of communication:

User location: determination of the end system tobe used for communication;

User capabilities: determination of the media and

media parameters to be used;

User availability: determination of the willingnessof the called party to engage in communications;

Call setup: "ringing", establishment of call

parameters at both called and calling party;

Call handling: including transfer and termination ofcalls.

SIP Functionality (cont.)


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SIP can also initiate multi-party calls using amultipoint control unit (MCU) or fully-

meshed interconnection instead of

multicast.

Internet telephony gateways that connect

Public Switched Telephone Network (PSTN)

parties can also use SIP to set up calls

between them.

Development of SIP


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SIP developed by Handley, Schulzrinne,Schooler, and Rosenberg

- Submitted as Internet-Draft 7/97

Assigned RFC 2543 in 3/99

Goals: Re-use of & Maximum Interoperabilitywith existing protocols

Alternative to ITUs H.323- H.323 used for IP Telephony since 1994

- Problems: No new services, addressing,features

- Concerns: scalability, extensibility

SIP Philosophy


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Internet Standard

- IETF - http://www.ietf.org Reuse Internet addressing (URLs, DNS, proxies)

- Utilizes rich Internet feature set

Reuse HTTP coding

- Text based Makes no assumptions about underlying protocol:

- TCP, UDP, X.25, frame, ATM, etc.

- Support of multicast

SIP Architecture


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SIP uses client/server architecture

Elements: SIP User Agents (SIP Phones)

SIP Servers (Proxy or Redirect - used to locate SIPusers or to forward messages.)

Can be stateless or stateful SIP Gateways:

To PSTN for telephony interworking

To H.323 for IP Telephony interworking

Client - originates message Server - responds to or forwards message

SIP Entities


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User Agents

User Agent Client (UAC): Initiates SIP requests

User Agent Server (UAS): Returns SIP responses

Network Servers (diff. types may be co-located )

Proxy: Decides next hop and forwards request, relays call

signaling , operates in a transactional manner, saves no

session state

Redirect: Sends address of next hop back to client,

redirects callers to other servers

- Registrar:Accepts REGISTER requests from clients,

maintains users whereabouts at a location server

SIP Operation


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1. SIP Addressing

2. Locating a SIP Server

3. Sending SIP Requests : SIP Transactions

4. SIP Methods

5. SIP Responses

6. Subsequent Requests and Responses

SIP t l

SIP protocol and mechanisms


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122

SIP protocolsSIP messages

Informational

Success

Redirection

Failure of Request

Server failure

Global failure

1xx

2xx

3xx

4xx

5xx

6xxRegister withLocation server

Features supported

Cancel pendingrequest

Release call

Confirm finalresponse

Initiate callINVITE

ACK

BYE

CANCEL

OPTIONS

REGISTER

Requests (Methods) Responses

from the client to the server from the server to the client

Step #1:SIP Addressing


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Uses Internet URLs

Uniform Resource Locators

Supports both Internet and PSTN addresses

General form is name@domain

To complete a call, needs to be resolved down to

User@Host

Examples:

sip:[email protected]

sip:J.T. Kirk

sip:[email protected];user=phone

sip:[email protected]

sip:[email protected];phone-context=VNET

Step#2: Locating a SIP server


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A caller first locates the appropriate server

When client wants to send a request URI

client will either send it to

- Locally configured Proxy server or to

- IP address & port corresponding to the

request URI [similar to the one in step#1]

Client must determine IP address, port of

server and the protocol to be used.

Locating (cont.)


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Client

1. Should try to contact a server at the port listed in

request URI. If no port specified then try port5060

- Use specified protocol if applicable

- o.w. use UDP if supported,

- if UDP fails or o.w. use TCP2. Send the request to the servers IP address if the

host part of request URI is an IP address o.w.

3. Find one or more address of server by querying

DNS,4. Results MAY be cached.

5. Capability to interpret ICMP messages must exist

Step# 3:Send a SIP request


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Once the host part has been resolved to a SIP

server, - client sends 1 / more SIP requests to thatserver &

- receives 1 / more responses from the server.

SIP Request-line (Messages) defined as:

Request-URI SIP-Version

(SP=Space, CRLF=Carriage Return and Line Feed)

(Method = INVITE | ACK | OPTIONS |

BYE | CANCEL | REGISTER)

Example:

INVITE sip:[email protected] SIP/2.0

Order of Operation


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Step1: Caller Issues Initial INVITE Request

Step 2: Callee Issues ResponseStep 3: Caller Receives Response to Initial

request

Step 4: Caller or Callee GenerateSubsequent

requests

Step 5: Receive Subsequent Requests

Step 6: BYE to end sessionStep x: CANCEL may be issued

Methods (cont.)


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INVITE: Initiates sessions

- Session description included in message body

Re-INVITEs used to change session state

ACK confirms session establishment, can only beused with INVITE

BYE terminates a session (hanging up)

CANCEL cancels a pending invite

REGISTER: binds a permanent address to currentlocation, may convey user data

OPTIONS: capability inquiry

Proxy Server Example


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Redirect Server Example


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SIP Responses


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SIP Responses defined as (HTTP-style):

SIP-Version SP Status-Code SP Reason-Phrase

CRLF

(SP=Space, CRLF=Carriage Return and Line Feed)

Example:SIP/2.0 404 Not Found

First digit gives Class of response

SIP Responses (cont.)

1xy Informational


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1xy Informational

request received , continuing to process request

2xy Successaction successfully recvd., understood & accepted

3xy Redirection

Further action to be taken to complete the request

4xy Client errorrequest contains syntax error or cant be completedat this server

5xy Server error

server fails to fulfill an apparently valid request 6xy global failure,

request is invalid at any server

Step #4 Generate Subsequent Requests


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Once the call has been established, either the calleror callee may generate INVITE or BYE requests tochange or terminate the call.

For the desired call leg the headers are set as follows(both including any tags):

- the To header field is set to the remote address, and- the From header field is set to the local address.

The Contact header field maybe different than theContact header field sent in a previous response orrequest. The Request-URI maybe set to the value of

the Contact header field received in a previousrequest or response from the remote party, or to thevalueof the remote address. [supports mobility]

SIP Requests Example


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Required Headers (fields):

Via: Shows route taken by request.

Call-ID: unique identifier generated by client.

CSeq: Command Sequence number

generated by client Incremented for each successive request

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP host.wcom.com:5060

From: Alan Johnston

To: Jean Luc Picard

Call-ID: [email protected]

CSeq: 1 INVITE

}Uniquely

identify

this

session

request

Via Field in Header


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The Request headers include a Via field

The Via field indicates the path taken by therequest so far.

Every proxy adds a Via Header with itsaddress to make sure that responses within

a transaction take the same path (to avoidloops, or to make sure that same firewallwill be hit on the way back)

This prevents request looping and ensures

replies take the same path as the requests,which assists in firewall traversal and otherunusual routing situations.

Via Headers and Routing


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Via headers are used for routing SIP messages

Requests Request initiator puts address in Viaheader

Servers check Viawith senders address, then add ownaddress, then forward. (if different, add receivedparameter)

Responses Response initiator copies request Viaheaders.

Servers check Viawith own address, then forward to nextViaaddress

All Via headers are copied from request to response

in order Response is sent to address in top Via header

Via Header (cont.)


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Step #5 Receiving Subsequent Requests S bseq ent to receipt the follo ing checks are made


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Subsequent to receipt, the following checks are made:

1. If the Call-ID is new,

- the request is for a new call, regardless of the valuesof the To and From header fields.

2. If the Call-ID exists,

- the request is for an existing call.

- If the To, From, Call-ID, and CSeq values exactly match(including tags) those of any requests received previously,

- the request is a retransmission.

3. If there was no match to the previous step,

- To & From fields compared against existing call leg local andremote addresses.

- If there is a match, & the CSeq in the request > last CSeq

received on that leg,- the request is a new transaction for an existing call leg.

Reliability


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If UDP is used:

-SIP client should retransmit a BYE, CANCEL, OPTIONS, orREGISTER request, exponential backoff, starting at a T1

second interval, doubling the interval for each packet, and

capping off at a T2 second interval.

-Retransmit a INVITE request with an interval that starts at T1

seconds, exponential back off, cease retransmissions if aprovisional or definitive response recvd., or once it has sent a

total of 7 request packets

Clients using TCP do not need to retransmit requests

Authentication & Encryption


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SIP supports a variety of approaches:

end to end encryption hop by hop encryption

Proxies can require authentication:

Responds to INVITEs with 407 Proxy-Authentication Required

Client re-INVITEs with Proxy-Authorizationheader.

SIP Users can require authentication:

Responds to INVITEs with 401 Unathorized

Client re-INVITEs with Authorizationheader

Features and Benefits


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Feature Benefits

InternetEnabled

SIP-based systems can take advantage of the growth ofthe Internet. Translating gateways permit SIP-based

systems to contact parties on the Public Switched

Telephone Network (PSTN) without being encumbered

by its legacy standards.Scalability Architecture permits inexpensive scaling. H/W & S/W

requirements for adding new users to SIP-based

systems is greatly reduced.

Simplicity SIP stack is smaller. SIP can be considered as a simpletoolkit that enables smart endpoints, gateways,

processes & clients to be built and implemented

Features and Benefits(cont.)

F B fi


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Feature Benefit

Distributed

functionality

- De-centralized intelligence permits more

functionality within each component. -

Changes made to specific components have a

minor impact on the rest of the system. It is

possible to connect one SIP phone to anotherwith an Ethernet cable & make calls between

the sets without the aid of any other server

modules.

- The other system components becomeuseful when the network requires more than

two phones.

SIP Summary


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SIP is:

Relatively easy to implement

Gaining vendor and carrier acceptance

Very flexible in service creation

Extensible and scaleable Appearing in products right now

SIP provides its own reliability mechanism& is therefore independent of the packet

layer and only requires an unreliabledatagram service

But


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SIP is not:

Going to make PSTN inter-working easy

Going to solve all IP Telephony issues

(QoS)

Secure !?SIP is still evolving and being extended as

technology matures and SIP products are

socialised in the marketplace.

Newer RFCs: #3261 through #3265

IP SIP Phones and Adaptors


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1

2

3

Are Internet hosts

Choice of application

Choice of server

IP appliance

Implementations

3Com (2)Cisco

Columbia University

Mediatrix (1)

Nortel (3)Pingtel

SIP protocols H.323 vs SIP

SIP and H323 Comparision


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146

H.323 vs SIP

H.323 SIP

Coding

QoS

Protocol

Complexity

ServerComplexity

Services

Specifications

Deployment

TerminalComplexity

Binary notation (ASN.1) Text-based (like HTTP)

RTP,RSVP, DiffServ RTP,RSVP, DiffServ

Multiple, circuit-oriented;

Version compatibility

Flexible with gateway

Gatekeepers are stateful

Basic call control;

IN services under study

ITU Standards

Widespread Limited

IETF RFC

Basic call control and routing;

IN services under study

SIP servers can be stateless

Moderate; terminal based

Intergrated with TCP/IP suites;

Feature compatibility

H.248/MEGACO Overview


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MEdia GAteway Control Protocol [RFC3015]

H.248 is ITU-T reference for the same protocol

Protocol for controlling telephony gateway and

terminals (IP Phones)

Basis for Vendor Independent Network

deployment

H.248/Megaco evolution

MGCP proposal by


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SGCP MGCP

Megaco/H.248

I-RFC 2705

MGCP released asInformational RFC

(Oct 99)

MGCP proposal bymerging IPDC and

SGCP

(Telcordia & Level 3)

Consensus between IETF andITU on Megaco Protocol

(March 99)

Lucent submits MDCPto ITU-T SG16(Nov 1999)

IETF

RFC 3015

MDCP(proposal)

IPDC

Megaco connection model

Based on 3 concepts:


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Based on 3 concepts:

Termination Identifies an end point for media flows Implements Signals, and generates Events Can appear in at most one context. Permanent (provisioned) terminations can exist

outside a context

Context

Defines communication between Terminations,acts as a mixing bridge

Contains 1 or more Terminations Supports multiple streams

Stream A context can have multiple streams, each

typically for a medium, e.g. audio, video, etc The MGC specifies which streams a given

termination supports

MG

Ta

Tb

Cn

Tc

Td

I2

O2=I1+I3

I1

I3

O3=I1+I2O1=I2+I3

Simple, powerful connection/resource model

Basic Concepts


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Megaco/H.248151

Connection model: terminations,

streams, and the context

Termination properties: descriptors

Message structure: transactions,

actions, and commands Event and signal processing

Packages

Terminations


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Megaco/H.248152

Source or sink of media flowsAvailable on both sides of the Media Gateway

(SCN and IP-Network)

Media flows can be either one way or two

way

Terminations can also sink/source

multimedia streams that include several

media streams.

Terminations


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Megaco/H.248153

Two types of terminations:

Persistent terminations: Instantiated by the

MG when it boots and remain active all the

time.

Ephemeral terminations: Created when theyare needed.

ROOTtermination: Represents the MG as an

entity in itself.

Context


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Megaco/H.248154

Modeled as a mixing bridge betweenterminations.

Two or more terminations may placed into a

context in order to mix and connect them.

Created and released by the MG undercommand of MGC.

Nullcontext: holds the persistent terminations

while they are not in use.

Context Example:

Basic call


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Megaco/H.248

155

Basic call

T1 T2

Medium=audio,

Mode=sendReceive

Medium=audio,

Mode=sendReceive

Ordinary two-party conversation between

terminations T1 and T2

Context Example:

Conversation with passive listener


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Megaco/H.248

156

Conversation with passive listener

T1 T2

Medium=audio,

Mode=sendReceive

Medium=audio,

Mode=sendReceive

T3

Medium=audio,Mode=sendOnly

Example:

Wiretap

Recording Machine

Context Example:

Media Conversion


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Megaco/H.248

157

Media Conversion

T1 T2

Medium=audio,

Mode=sendReceive

Medium=text,

Mode=sendReceive

Implicit conversion between media

Context Example:

M lti di


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Megaco/H.248

158

Multimedia

Stream=1,

medium=audio

Stream=1,

medium=audio

Stream=2,

medium=videoStream=2,

medium=video

T3

T1 T2

Descriptors


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Megaco/H.248

159

Properties of terminations

Most important ones:

Media Descriptor : Describes the

transformations to be applied to media flows

through the termination

Events Descriptor: Selects and reports events

that are currently occurred and important for

MGC

Signals Descriptor : Indicates which signals the

MGC currently wishes the MG to play out thetermination

Descriptors


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Megaco/H.248

160

Media Descriptors include some other descriptors: Termination State Descriptor: Carries the state of the

termination which is independent of any media flow

Local Control Descriptor: Provides Media-stream-

related information relevant only between MGC and MG

Local and Remote Descriptor: Carries information

describing media flows within a stream which must be

coordinated with the remote entity

Message Structure


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Megaco/H.248

161

Megaco/H.248 message

Header Transaction Transaction ... Transaction

Req or Reply Req or Reply Req or Reply

Trans Hdr Action ... Action

Ctx Hdr Ctx Properties Command ... Command

Cmd Hdr Descriptor ... Descriptor

Commands


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Megaco/H.248

162

Megaco uses some commands in order tomanipulate terminations, contexts, signals and

events.

For termination manipulation: Add, Subtract,

Move, Modify

For event reporting: Notify

For management: AuditCapability, AuditValue,

ServiceChange

Commands


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Megaco/H.248

163

From MGC to MC:

Add: adds a termination to a context

Subtract: removes a termination from a context

Move: moves a termination from a context to

another

Modify: changes the characteristics of an existing

termination , which can be in the null context

AuditValue & AuditCapabilities: return information

about terminations, contexts and general state and

capabilities of MG

Commands


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Megaco/H.248

164

From MG to MGC:

Notify:MG sends it to inform MGC that an

event has occured.

Either from MG to MGC or from MGC to MG:

ServiceChange:creates a connectionbetween MG and MGC.

Descriptors are parameters for all these

commands & return values of some of them.

Events

Events are detected at MG and reported to


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Megaco/H.248

165

Events are detected at MG and reported to

MGC.(example: inband signaling)

MGC controls what events it wants to learnabout at any given time

sets the termination Events descriptorEvents can have side effects

stop play out of signals

start new signals automatically update the set of events of

interest

Signals


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Megaco/H.248

166

Signals cause things to happen onterminations

play a tone, display text, ...

Specified in the Signals descriptor for a

terminationMGC can specify duration of signal ahead of

time or signal can play until explicitly stopped

Signals stop playing when any event is

detected unless MGC says otherwise.

Megaco/H.248: Commands


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Provides control for manipulating terminations and contexts.

Current Command Set:

Command Initiator Description

Add MGC Adds a termination to a context.

Modify MGC Modifies a terminations properties, events, and

signals.

Move MGC Moves a termination from one context to another.Subtract MGC Removes a termination from its context.

AuditValue MGC Returns current state of properties, events, signals,

and statistics.

AuditCapabilities MGC Returns all possible values for termination

properties, events, and signals allowed by an MG.

Notify MG Informs MGC of event occurrence(s).

ServiceChange MGC Takes or places a termination(s) out of or in service.

MG For registration and restart; notifies MGC

termination(s) will be taken out of or returned to

service.

Signalling in Mobile Networks


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Cellular and broadband wireless Technology Evolution


Signaling in GSM


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g g

MSC - Mobile Services Switching Centre

BSS - Base Station System

HLR - Home Location Register

VLR - Visitor Location Register

EIR - Equipment Identity Register

MS - Mobile Station

SSP - Service Signalling Point in SS7 Network


Signaling in GSM


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Signalling in Mobile NetworksOMAP - Operations Maintenance

and Administration Part Non-Circuit related Circuit related

SS7 Protocol ModelOSI Model


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ASE - Application Service

Element

TCAP - Transaction Capabilities

Application Part

ISDN-UP - ISDN User Part

SCCP - Signalling Connection

Control Part

MTP - Message Transfer Part

ISDN-UP

SCCP

MTP Level 3

MTP Level 2

MTP Level 1

Network

Data Link

Physical

Presentation

Session

Transport

Application

User

Parts

Transport

Services

Parts

Null

TCAP

OMAP ASEs


Signaling in GSM


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Location registration and cancellation

Handover procedures

Handling of supplementary services

Retrieval of subscriber parameters during call set-up

Authentication procedures

OA&M

MTP L1

MTP L2

MTP L3

SCCP

TCAP

ASEsOMAP

Mobile Application PartMAP

Mobile

Application

Entity (AE)


SCP S i C l P i


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HLR

BSSBSS

VLR SSF

MSC

SCP

BSSBSS

VLRSSF

MSC

MAPMAP

HLR/VLRsManage

Mobility

SCP Manages

service aspects

Eg. Pre-paid accounts

SCP - Service Control Point

SSF - Service Switching Function

INAP - Intelligent Network ApplicationPart of C7

MAP - Mobile Application Part of C7

IN Implementation in a GSM Network


The Intelligent Network


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Charging options

Freephone

Local rates

Premium rates

Routing options

Group calls

Network call centre options

Virtual Private Networks

Personal calls

one number configurable routing

The IN adds a computer

and voice system to the

fixed network

Service Control

Point

Intelligent

Peripheral

g


BSS/MSC Interface (A)


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BSSAP - BSS Application PartDTAP - Direct Transfer Application Part

BSSMAP - BSS Management

Application Part

BSSOMAP - Operations and Maintenance

Application Part

SCCP - Signalling Connection Control

Part


MSC BSS

A

MTP Level 3

MTP Level 2

MTP Level 1

Distribution

BSSMAPDTAP

BSSOMAP

SCCP

BSSAP

BSS/MSC Interface (A)


A-Interface to MSC


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0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31

SIGNALING

SYNC

Voice and/or Data Voice and/or Data

One PCM 30 Frame

GSM A-Interface

Signalling Protocol Stack

Connectivity Management

Mobility Management

BSS Application PartSignalling Connect Control Part

Message Transport PartLayer 3




Abis


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BTS BSCAbis

Physical Layer

Distribution

Non

TransportTransport

LAPD

Traffic Management

Procedures

Network

Management

L2

Management

Radio Link

DistributionOther Processes

L1

L2

L3


Um


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CC - Call Control entity

SS - Supplementary Services support entity

SMS - Short Message Service support entity

LAPDm - Link Access Procedures on Dm

channel (Similar to ISDN LAPD)

Physical Layer

(RADIO SUBSYSTEM)

LAPDm

L1

L2

Mobility Management (MM)

Connection Management (CM)

L3

Radio Resource Management (RR)

SMSSSCC

MS

m


MSBTSBSCMSC Signalling Protocol Model


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MS BTS BSC MSC

MS

MM

LAPDm

Phys.

RR

LAPDm

Phys.

LAPDm

Phys.

RR BTSM

LAPD

Phys.

SCCP

MTP

RR BSSAP

A

MM

BTSM SCCP

MTP

BSSAP

A-bisUm

CM

CM - Connection Management

MM - Mobility Management

RR - Radio Resource Management

LAPD - Link Access Procedures on D channel

LAPDm - Link Access Procedures on Dm channel

BTSM - BTS Management

BSSAP - BSS Application Part

SCCP - Signalling Connection Control Part



UMTS Rel 6 7

Documents

Baohieu Va Dieukhien Ketnoi