PSTN
Public Switched Telephone Network
NGN & PLMN
Key parts of telecommunication network
Trunk Network
1. Access network
Node 1 Node 2
2. Switch/Node system Access Access
3. Trunk transmission system Node 3
4. Signaling Terminals Terminals
5. Rating, Billing and Provisioning system
Note – Power system is key but not discussed here.
Signalling
Mechanism that allows
Network entities
Customer premised devices
Network switches
To establish , maintain and terminate sessions in a network
Telephone
Is a device which converts human speech in the form of sound
waves produced by the vocal cord to electrical signals.
These signals are then transmitted over telephone wires and then
converted back to sound waves for human ears.
Microphone
Earphone
Signaling functions
POTS telephone instrument
Speaker diaphram Receiver
(moveable) (earpiece)
Sound
Waves
Handset
Transmitter
Sound
(mouthpiece) RJ-11
Waves
connectors
RJ-22 connector 2 wires
4 wire RJ-22
connector
Basic analog telephone
Exchange Line Phone Instrument
Idle Active
Hook switch open Hook switch close
Only Ringer works for an AC Ringer circuit by-passed by hook
current switch
Speech path is open DC current flows in the line
No DC current flows in the line Speech path is operational
Exchange telephone line interface
ring switch
Telephone T
tip (+)
TX
Subscriber current hybrid
Line ring (-) RX
detector
control channel
ring
generator -48 VDC Processor
~
(100Vrms 25 Hz)
POTS telephone instrument
Pulse dialing
Subscriber and exchange signaling
On/off hook DC current flow
Dial-tone 350Hz + 440 Hz continuous
Pulse Dialing / DTMF
L1 697 L2 770 L3 852 L4 941
H1 1209 H2 1336 H3 1477 H4 1633
Ring AC with cadence
Ring-back 440Hz + 480Hz with cadence
Busy 480Hz + 620Hz with 1/2:1/2 cadence
Receiver off-hook
Subscriber Signaling
Types of signaling
Telephone to Exchange Exchange to Exchange
Analog voltage signaling Register Signaling
loop-start, ground-start, E&M MFR1, MFR2
Channel associated signaling
DTMF signaling (CAS)
Common channel signaling
(CCS) SS7
CAS signaling
signaling is sent over the same channel over which voice calls
are carried.
Signaling equipment needed for every trunk
Slow to operate
Once call is established no control functions possible
Failure of trunk has no impact on whole system
v+s
v+s
Ex1 v+s Ex 2
v+s
CCS signaling
Voice trunks used when only connection is needed
Out of band signaling
Needs dedicated separate path for signaling
Control functions possible while call progresses
Failure of signaling links has high impact on whole system
v
v
Ex1 v Ex 2
s
CCS signaling / SS7
Signaling is a network itself and is the key controller.
All nodes are called “Signaling / Service Points”
Service Control Points (SCP)
Service Switching Points (SSP)
Signaling Transfer Points (STP)
Ex Ex
1 2
SSP SSP
Ex
3
STP
STP SSP
SCP
(1) Service Switching Points (SSP)
These are telephone exchanges with local subs connected
Have SS7 interfaces.
Set up , Manage, and release voice circuits
Communicate via ISUP and TCAP protocols
(2) Signaling Transfer Points (STP)
Routers or gateways in the signaling network
Messages are not originated by them
Provides destination address of the SSP to originating
SSP
Interface with other networks and offer protocol
conversion
Provides traffic and usage measurements
(3) Service Control Points (SCP)
Provides application access
Is a interface to a database
Exchange (SSP) , STP, SCP
interworking
ISUP Messages
SSP STP
IAM = Initial Address Message
ACM = Address Complete Message
ANM = ANswer Message
REL = RElease Message
RLC = Release Complete Message
Telephone voice/signaling network
Class 4
Tandem Switch Class 5
End Office Switch
PSTN / Circuit Switched Network
SS7 Signaling
ISUP Messages
Signaling Control Layer
Signal Transfer Points
Voice Transport Layer
Class 4
Tandem Switch Class 5
End Office Switch
PSTN / Circuit Switched Network
SS7 Signaling
Service Control Point
ISUP Messages
INAP/ TCAP Messages
Signal Transfer Points
Voice Transport Layer
Class 4
Tandem Switch Class 5
End Office Switch
PSTN / Circuit Switched Network
EX2
EX1
STP STP
1 2 SSP
SSP
2
1
Off hook
Dial Tone
Dialing IAM
IAM IAM
ACM ACM ACM
Ring back Ringing
Off hook
ANM ANM ANM
conversation
REL REL REL Hang up
Hang Up
RLC RLC RLC
When a call is placed to an out-of-switch number,
the originating SSP1 transmits an ISUP Initial Address
Message (IAM) to reserve a trunk circuit from the originating
1 SSP1 to the destination SSP2.
The IAM is routed via the local STP1 of the originating SSP1 to
the destination SSP2
The destination SSP2 determines if it serves the called party.
If so it generates a ringing tone at the called party's line and
transmits an ISUP Address Complete Message (ACM) to the
2 originating SSP1 via its local STP2.
The ACM indicates that the remote end trunk circuit has been
reserved.
STP2 routes the ACM to SSP1 which generates a ringing tone
to the calling party's line and connects it to the trunk circuit.
When the called party picks up the phone, SSP2 terminates the
ringing tone and transmits an ISUP ANswer Message (ANM)
to SSP1 via STP2.
3 STP2 routes the ANM to SSP1 which verifies that the calling
party line is connected to the reserved trunk and if so starts
billing
If the caller hangs up first SSP1 sends an ISUP RELease
message (REL) to release the trunk circuit between the 2
switches.
4A STP1 routes the REL to SSP2.
Upon receiving the REL SSP2 disconnects the circuit from the
called party's line and transmits an ISUP ReLease Complete
message (RLC) to SSP1 to ack the release of the trunk circuit.
When SSP1 receives the RLC it terminates billing.
If the called person hangs up first, SSP2 sends a REL to SSP1
indicating the release cause.
When REL is received, SSP1 disconnects the circuit from the
4B caller's line and transmits an ISUP ReLease Complete
message (RLC) to SSP2.
When SSP2 receives the RLC it terminates and stops billing.
CCS / SS7 facilitates two switching paradigms
[Link] switching for voice
[Link] switching for signaling
A circuit is held throughout out a call
Signaling goes as data packets during initiation and
completion of the call.
There may be occasional data packet sharing in case caller
and callee require enhance features such as conferencing,
SS7 protocol stack
OSI
7 Layer
model
MTP = Message Transfer Part
SCCP = Signaling Connection Control Part
TCAP = Transaction Capabilities Application Part
MAP = Mobile Application Part
INAP = Intelligent Network Application Part
ISUP = ISDN User Part
SS7 stack revisit
OSI Layers
Application INAP MAP
Presentation
TCAP ISUP
Session
Transport SCCP
Network MTP Level 3
Data Link MTP Level 2
Physical MTP Level 1
Signaling Connection Control Part (SCCP)
Transactional Capabilities Application Part (TCAP)
Intelligent Network Application Part (INAP)
Mobile Application Part (MAP)
ISUP msg formats
Service Indicator (isup/tcap..)
Destination Station Code
Originating Station Code
Circuit Allocated
Msg type Message direction Content
Message type
01 IAM Forward Service type (voice /
data)
Optional data fields Called party number ,
calling party number
Calling party name
06 ACM Backward
09 ANM Backward
12 REL Both Release code
16 RLC Both
CAS and CCS comparison
CAS CCS
Trunks must be held during signaling Trunks not required during signaling
Scope is limited (working with a Extensive scope is possible
database is very hard) (enhanced services, IN services etc)
Interference between voice and No interference between voice and
signaling may possible signaling
Signaling equipment needed for all Only one equipment is required for
trunks group of trunks
Potential misuse by customers who can Signaling channels cannot be accessed
mmic signaling possible by users
Signaling is comparatively slow Signaling is significantly fast
Speech circuit reliability is guaranteed Speech circuit quality cannot be
guaranteed
Changes are hard as all trunks needed Signaling can be added modified easily
to be changed
No signaling during speech is possible Signaling during speech time is
possible
Schematic – telephone exchange
Subscriber Switching system
stage
LIC switch Trunk
1 links to other
network
Trunk
LIC elements
2
Tone Tone Rx Signaling
Line generator .
interface
circuit • Switch control
• E.164 number analysis
SS7 Signalling
• Charging equipment
• User databases
• O&M functions Control system
Telephone network architecture
Local
Exchang
e
Master
Exchang
Trunks e
2
Master
Local Exchang
Exchang e
e 1
Master
Local loops
Exchang
e
3
Local
Exchang Local
e Exchang
e
Telephone numbering
Local 121 D
Exchang
e
Master
121 Trunks
Exchang 122
e 38
2
A 11 123
2 Master
Local Exchang
B Exchang e
e 1
122 C Local loops
Master
45 Exchang
123 5
e
3
Local
Exchang Local
e Exchang
e
F
121 E
123 121 123
122
122
VOIP – voice over IP
Gathering momentum in business and consumer segments
1. very much cost effective for international / long distance calling
2. works on almost all popular devices (PC, laptops, smart phones
etc)
3. Ability to make video calls and conferencing
4. no need special infrastructure
(public internet or intranet is enough)
5. Offers new business avenues / Threat to legacy telecom
operators
Sip – session initiation protocol
SIP – the most important protocol used in VOIP
Operates on Application layer
Facilitate sending voice , multimedia, video over the IP based
media.
Highly flexible
Sip – key points
Is a signaling protocol responsible for creating , modifying and
termination a multimedia session over ip
Described by IETF rfc3261
Use following companion protocols
SDP session description protocol
Describe the session , media etc
RTP real time protocol / RTCP
Delivering voice and video on real time basis.
Embodies client -server architecture
URL/URI headers and text encoding as HTTP
Following SMTP header style
Sip – network elements
User agent
Proxy server
Register server
Redirect server
Location server
Sip – network elements
User agent
Proxy server
Registrar server
Redirect server
Location server
Location server
4 query
2 registration
proxy server
registrar
3 invite 5 invite
1 registration
User agent User agent
Sip – User agent
End point of a SIP network
Most of the cases a customer device
(phone, soft phone, laptop, smart phone)
Can initiate , modify or terminate a session
Logically divided in to two parts
user agent client (uac) -sends request receives response
user agent server (uas) -sends response receives requests
User agent address “uri” looks like
“sip:user@[Link]”
Sip – proxy server
Receives requests and forwards to other another sip element
Acts like a router
Understands the request by reading the uri and send it ahead
Sits in between two users
Limited a max of 70 between a source and a destination
Sip – register server
Accepts registration requests from user agents
Allows user agents to authenticate themselves to join a voip
network
Stores the uri and locations of users in a database.
Registration entry contains :
user_agent sip address
user_agent IP address
user agent needs to repeatedly refresh the registration.
Helps other sip servers in the network to trace end point
Sip – redirect servers
redirects the request back to the client
Indicate client needs to try a different route
happens when a recipient has moved from its original position
(temporarily or permanently)
Sip – location servers
Stored addresses registered to a Registrar
Sip – network elements
Location server
4 query registration
2
proxy server
registrar
3 invite 5 invite
1 registration
User agent User agent
SIP transactions
SIP transaction consists of a request and one or more replies.
The most commonly “Requests”: REGISTER, INVITE and BYE.
Common “Replies” are:
ACK, and status replies contain 3-digit status code and a human-
readable text
100 Trying 400 Bad Request
180 Ringing 401 Unauthorized
200 OK 404 Not Found
301 Moved Permanently 500 Server Internal Error
600 Busy Everywhere
302 Moved Temporarily
603 Decline
Sip – flow of actions
Registration
Call Set up
Call
progress
Termination
ISUP SIP inter-working thru gateways
EX EX
1
STP1 GW Internet GW STP2 2
Dial digits IAM Invite
100 Trying IAM
Ringing
ACM
Ring Tone ACM 180 Ringing
ANM 200 OK ANM Off hook
Voice
Hang up REL Bye REL
200 OK Hang up
RLC RLC
