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Group 4 - PSTN

The document provides an overview of the Public Switched Telephone Network (PSTN), detailing its function as a global circuit-switched network for voice communication and its evolution from the late 19th century to modern times. It discusses core components such as end-user devices, switching centers, and transmission media, as well as signaling systems used for call connections. The report emphasizes the importance of PSTN in providing reliable and high-quality voice communication, while also integrating with modern technologies like VoIP.

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0% found this document useful (0 votes)
12 views28 pages

Group 4 - PSTN

The document provides an overview of the Public Switched Telephone Network (PSTN), detailing its function as a global circuit-switched network for voice communication and its evolution from the late 19th century to modern times. It discusses core components such as end-user devices, switching centers, and transmission media, as well as signaling systems used for call connections. The report emphasizes the importance of PSTN in providing reliable and high-quality voice communication, while also integrating with modern technologies like VoIP.

Uploaded by

Donna Belle
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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Group 4 Report – PSTN

Zaki Jeremy A. Albaran

Renzelle Anjelo P. Esguerra

I. Introduction to PSTN

A. What is PSTN?

• A global circuit-switched network for voice communication


o The Public Switched Telephone Network (PSTN) is a worldwide infrastructure
that enables voice calls through a circuit-switched system.
o Calls are routed through a network of telephone exchanges, copper wires,
fiber optics, and satellite links.
• Used for landline telephone services
o PSTN is the foundation of traditional fixed-line telephony, connecting homes
and businesses via wired connections.
o Unlike mobile networks, PSTN primarily relies on physical infrastructure like
telephone poles, underground cables, and exchange centers.
• Provides reliable, long-distance voice transmission
o PSTN ensures high call quality and minimal signal degradation, even over
long distances.
o The network is designed for uninterrupted voice communication, making it
ideal for emergency services, government institutions, and businesses.

Purpose of PSTN

• Ensures high-quality, stable voice communication


o PSTN is engineered to prioritize voice clarity and reliability, avoiding the
latency and packet loss often seen in digital networks.
o It provides a consistent connection that does not depend on internet
bandwidth, making it more stable for critical communications.
• Supports both analog and digital telephone systems
o While originally designed for analog signals, PSTN has evolved to
accommodate digital communication through ISDN and PCM technology.
o This hybrid approach allows for compatibility with older telephone systems
and newer digital infrastructure.
• Integrates with modern networks like VoIP and DSL
o PSTN is still used today alongside modern technologies like Voice over IP
(VoIP) and Digital Subscriber Line (DSL) for internet access.
o Many businesses use VoIP gateways to connect PSTN with cloud-based
communication systems, ensuring a gradual transition to digital telephony.
B. Evolution of PSTN

The Public Switched Telephone Network traces its roots back to the late 19th century. In
1876, Alexander Graham Bell was awarded the patent for his invention of the telephone,
laying the foundation for the historical phone systems we know and love.

In the early days, the telephone was a novelty, mostly used for one-on-one communication
between two phones connected by a direct analog telephone line. If you wanted to call
people in other locations, you would need multiple phones that were physically connected
by wire to each of those places.

Eventually, as the popularity of the telephone grew, so did the need for a more advanced
system that could handle multiple connections at the same time. This led to the
establishment of telephone exchanges, which acted as intermediaries by having operators
connect calls manually through a switchboard.

The first telephone exchange opened in 1878 in New Haven, Connecticut, and connected
21 clients, including the local police, post office and a drug store. By the 1890s, exchanges
were being established in cities across the United States and Europe, creating a patchwork
of local phone networks that were gradually being linked together. Thus, the Public
Switched Telephone Network was born.

By the early 1900s, the PSTN had expanded significantly. In the 1930s, automatic
electromechanical switching technology allowed those with rotary phones to locally dial
each other without the help of an operator. While this new technology caused the number
of human operators to dwindle, it expanded the network’s capacity. and let more people
access telephone services. The process of connecting voice calls was faster and more
efficient.

In those days, the PSTN only supported voice communication in the form of basic phone
calls. Long-distance calls were expensive and required a human operator and scheduling
in advance. Still, it was the beginning of a communication revolution. The telephone quickly
became an important device in everyday life, linking families, businesses and communities
in ways that were previously unimaginable.
In the decades after its establishment, the Public Switched Telephone Network served as a
vital communication link. At the same time, advancements in technology would regularly
influence and alter how the telecommunication systems operated.

For example, up until the 1950s, the PSTN relied mostly on analog technology, which
transmitted voice signals over copper wires and relied on electromechanical switches.
However, the introduction of electronic switching systems (ESS) in the 1960s marked a
major shift because they routed calls with digital, not analog, signals. This innovation led to
improved call performance and features like call waiting, call forwarding, speed dialing and
conference calls.

Other technologies also expanded the PSTN’s capabilities. Long-distance microwave


transmission systems went digital in the 1980s, enabling the support of a wider range of
services over the PSTN. Meanwhile, fiber optics played an important role in enabling faster
data rates and improved reliability. Cellular networks also became part of this ecosystem,
further expanding the reach of the PSTN.

The 1980s saw the introduction of the Integrated Services Digital Network (ISDN), which
allowed the PSTN to handle not just voice but also data, video and other network services
over the same lines. This would pave the way for services like faxing and early internet
connections and was the start of the PSTN’s role in supporting more than just voice
communication.

By the 1990s, the internet was revolutionizing communication and quickly overtaking
traditional analog and ISDN in relevancy. Although initially separate from the PSTN, internet
technologies eventually connected with traditional phone systems, allowing for services
like dial-up internet and later broadband to use the PSTN’s infrastructure.

The rise of Voice over Internet Protocol (VoIP) in the early 2000s was another game-
changer. VoIP technology, which allows voice communication over the internet, began to
gradually replace traditional copper landlines for many businesses and households. With
crystal-clear calls now able to be transmitted digitally over IP networks, internet phone
service transformed the landscape, making telephony faster, cheaper and more reliable.

II. Core Components of PSTN

A. End Users
• Telephones - In the Public Switched Telephone Network (PSTN), both analog and
digital telephones serve as end-user devices for voice communication. Analog
phones transmit voice as continuous electrical signals over circuit-switched
networks, offering simplicity but suffering from signal degradation and interference.
In contrast, digital phones convert voice into binary data, improving signal quality,
efficiency, and features like caller ID and encryption. With technologies like ISDN
and VoIP, digital telephony enables advanced functionalities and better integration
with computer networks. While analog systems still exist, digital telephony has
largely replaced them, ensuring higher-quality communication and greater
efficiency.

• Private Branch Exchange (PBX) - A Private Branch Exchange (PBX) is a telephone


system used by businesses and organizations to manage internal and external
communications within the Public Switched Telephone Network (PSTN). As an end-
user device in the PSTN, a PBX acts as an intermediary between an organization's
internal phone network and the public network, allowing multiple users to share
external lines efficiently. It enables features such as call routing, voicemail, call
conferencing, and extension dialing, reducing the need for separate phone lines for
each user. Traditional PBX systems use analog or digital connections, while modern
IP-PBX systems leverage internet-based protocols for greater flexibility and
scalability. PBX systems enhance communication by optimizing call management
and reducing costs through efficient line sharing. Businesses can customize call
handling rules, implement automated attendants, and integrate with computer-
based applications for improved functionality. With advancements in VoIP and
cloud-based PBX solutions, organizations can now access even more features, such
as remote connectivity, mobile integration, and enhanced security. While traditional
PBX systems remain in use, the transition to digital and cloud-based solutions
continues to shape the future of business communication within the PSTN.

• VoIP Gateway - A VoIP gateway serves as an essential end-user device in the Public
Switched Telephone Network (PSTN), bridging the gap between traditional
telephone networks and modern Voice over Internet Protocol (VoIP) systems. It
enables seamless communication between analog or digital telephony systems and
IP-based networks by converting voice signals between circuit-switched and
packet-switched formats. This functionality allows businesses and service providers
to integrate VoIP services with existing PSTN infrastructure, reducing costs while
maintaining compatibility with traditional telephony. A VoIP gateway works by
converting analog voice signals from PSTN into digital packets for transmission over
an IP network and vice versa. When a VoIP user dials a PSTN number, the gateway
receives the digital VoIP packets, decompresses them, and converts them back into
an analog or digital signal before routing the call through the traditional phone
network. Similarly, for inbound calls from the PSTN to a VoIP system, the gateway
digitizes the voice signal, compresses it, and transmits it over the internet.
Advanced VoIP gateways also support additional features like call routing,
encryption, and traffic management to optimize voice quality.
B. Switching Centers
• Local Exchange - A local exchange in the Public Switched Telephone Network (PSTN)
is a central switching facility that connects telephone subscribers within a specific
geographic area. It serves as an intermediary between end-user devices, such as
landlines or PBX systems, and the broader telephone network, enabling local and
long-distance call routing. When a user makes a call, the local exchange processes
the dialed number, establishes a connection, and directs the call to the appropriate
destination, either within the same exchange or to another exchange via trunk lines.
Local exchanges also handle essential services like call forwarding, voicemail, and
emergency dialing. With advancements in telecommunications, traditional circuit-
switched local exchanges are gradually being replaced by digital and IP-based
systems, improving call efficiency and network scalability.

• Tandem/Toll Exchange - A tandem exchange in the Public Switched Telephone


Network (PSTN) is a switching center that interconnects multiple local exchanges,
facilitating call routing between different regions without directly connecting to end-
user devices. When a call is made between subscribers served by different local
exchanges, the tandem exchange acts as an intermediary, optimizing network
efficiency by reducing the need for direct connections between every local
exchange. This hierarchical structure helps manage high call volumes, prevents
network congestion, and ensures seamless communication over long distances. As
telecommunication networks evolve, tandem exchanges are increasingly integrated
with digital and IP-based systems, enhancing call routing speed, reliability, and
scalability.

• International Gateway - An international gateway in the Public Switched Telephone


Network (PSTN) is a specialized exchange that facilitates voice and data
communication between different countries. It serves as the primary access point
for international calls, converting signals between different telecommunication
standards and ensuring seamless connectivity across global networks. When a user
dials an international number, the call is routed from the local exchange to a
tandem exchange and then to the international gateway, which directs it to the
destination country’s network. These gateways handle protocol conversions, signal
processing, and regulatory compliance, ensuring clear and secure communication
across borders. With the rise of digital and VoIP technologies, international
gateways now integrate advanced data compression and encryption techniques,
enhancing call quality and reducing costs for global communication.

PSTN Topology

C. Transmission Medias
• Copper Wire - Copper wires have long been the backbone of the Public Switched
Telephone Network (PSTN), serving as the primary medium for transmitting voice
signals over traditional landline connections. These wires carry analog or digital
signals through electrical pulses, enabling communication between telephone
exchanges and end-user devices. Copper wiring is reliable, easy to install, and
provides consistent voice quality, but it has limitations such as signal degradation
over long distances and susceptibility to electromagnetic interference. With
advancements in telecommunications, many networks are transitioning to fiber-
optic cables and wireless technologies, which offer higher bandwidth and improved
efficiency. However, copper wires remain in use, especially in legacy systems and
rural areas where upgrading to newer infrastructure is still in progress.

• Fiber Optic Cables - Fiber optic cables play a crucial role in the modern Public
Switched Telephone Network (PSTN) by transmitting voice and data signals using
light pulses instead of electrical signals. Unlike traditional copper wires, fiber optics
offer significantly higher bandwidth, faster data transmission, and greater
resistance to signal degradation and electromagnetic interference. These cables are
commonly used for long-distance communication, connecting central offices,
tandem exchanges, and international gateways to ensure high-quality and reliable
connectivity. As telecommunications networks evolve, fiber optics are increasingly
replacing copper infrastructure, enabling advanced digital services, VoIP
communication, and high-speed internet alongside traditional voice services. Their
durability and efficiency make them essential for the continued expansion and
modernization of global telecommunication networks.

• A fiber microwave link in the Public Switched Telephone Network (PSTN) is a hybrid
communication system that combines fiber-optic and microwave technologies to
transmit voice and data signals. This setup is commonly used in areas where laying
fiber cables is challenging, such as remote or mountainous regions. The fiber-optic
segment ensures high-speed, long-distance transmission with minimal signal loss,
while the microwave link provides wireless connectivity between network points
where physical cabling is impractical. Microwave signals travel through the air using
line-of-sight transmission between antennas, enabling rapid deployment and
flexible network expansion. By integrating fiber and microwave technologies,
telecommunication providers can enhance network reliability, extend coverage, and
maintain efficient voice and data communication within the PSTN.

III. Signaling Systems and Call Connection Process

A. Overview of PSTN Signaling

Signaling is the process of sending control information for call setup and teardown. It can
be classified as:

• In-band Signaling: Uses the same channel as voice communication.

In-band signaling is used in land-line telephones to transmit control tones that route
calls.

In-band signaling was used in older telephone carrier systems to route calls.
In-band signaling can be heard by participants in a phone call.

• Out-of-band Signaling: Uses separate channels for control signals, improving


reliability.

Out-of-band signaling uses a separate channel, or signaling link, to carry control


signals.

This channel is used to establish and terminate connections, control flow, and
transmit error information.

Out-of-band signaling can improve call setup and teardown times, and reduce
interference between control signals and voice traffic.
B. Common Signaling Systems Used in PSTN

1. Signaling System No. 7 (SS7)

• A global standard for call setup, routing, and management.

• Uses out-of-band signaling for faster and secure communication.


2. Dual-Tone Multi-Frequency (DTMF) Signaling

• Used in telephone keypads for dialing.

• Generates specific frequency pairs for each digit pressed.


3. Common Channel Signaling (CCS)

• Uses a dedicated signaling channel, independent of voice transmission.

• Enables advanced call features like call forwarding and caller ID.

4. ISDN User Part (ISUP)

ISUP information is carried in the service information field (SIF) of an MSU. The SIF contains
the routing label followed by a 14-bit (ANSI) or 12-bit (ITU) circuit identification code (CIC).
The CIC indicates the trunk circuit reserved by the originating switch to carry the call. The
CIC is followed by the message type field - IAM, ACM, ANM, REL, RLC - which defines the
contents of the remainder of the message.
Basic ISUP Signaling

ISUP Message Format


IAM

An initial address message (IAM) is sent in the "forward" direction by each switch in the
circuit between the calling party and the destination switch of the called party. An IAM
contains the called party number in the mandatory variable part and may contain the
calling party name and number in the optional part

IAM Format

ACM

ACM An address complete message (ACM) is sent in the "backward" direction to indicate
that the remote end of a trunk circuit has been reserved. The originating switch responds to
an ACM message by connecting the calling party's line to the trunk to complete the voice
circuit from the calling party to the called party. The calling party hears ringing on the voice
trunk.

ACM Format

REL

REL A release message (REL) is sent in either direction indicating that the circuit is being
released due to a specified cause indicator. An REL is sent when either calling or called
party hangs up the call (cause = 16). An REL is also sent back to the calling party if the
called party is busy (cause = 17).
REL FORMAT

RLC

RLC A release complete message (RLC) is sent in the opposite direction of an REL to
acknowledge the release of the remote end of a trunk circuit and to end the billing cycle, if
appropriate.

RLC FORMAT
Signaling Comparisons

CCS
DTMF (Dual-
ISUP (ISDN SS7 (Signaling System (Common
Feature Tone Multi-
User Part) No. 7) Channel
Frequency)
Signaling)

Out-of-band
Type of Out-of-band signaling Out-of-band In-band
signaling (part
Signaling system signaling signaling
of SS7)

Controls call Call setup and Dialing


Provides full signaling
setup, control numbers and
Primary Use framework for telecom
teardown, and (precursor to interacting with
networks
management SS7) IVR systems

Moderate
Fast, since it
Very fast, as it avoids (better than Slower, since it
uses separate
Call Setup Time congesting voice in-band, but shares the
signaling
channels less efficient voice channel
channels
than SS7)

Less secure,
Secure due to More secure
vulnerable to
separate High security, reduces than in-band
Security spoofing and
control fraud risk methods, but
tone
signaling outdated
manipulation

Supports
advanced Highly flexible, Only used for
Limited to
telecom supports global basic dialing
Flexibility older PSTN
services (call telecommunications and menu
applications
forwarding, and mobile networks navigation
caller ID)

Used within Used in PSTN


Used worldwide in Used in
SS7 networks before SS7
Implementation modern PSTN, VoIP, landline
for call became
and mobile networks phones and
processing dominant
CCS
DTMF (Dual-
ISUP (ISDN SS7 (Signaling System (Common
Feature Tone Multi-
User Part) No. 7) Channel
Frequency)
Signaling)

automated
phone systems

Older PSTN
PSTN and ISDN Global telecom Phone dialing,
Example networks
call control, networks, mobile ATM PIN input,
Application before SS7
VoIP gateways networks, VoIP IVR systems
adoption

C. Call Connection Process in PSTN

Call Initiation
Call Answered
Call Release
Steps

1. Calling party goes "off hook" on an originating switch (SSP) and dials the directory
number of the called party.

1a. Originating SSP transmits ISUP IAM to reserve an idle trunk circuit. The IAM includes
OPC, DPC, CIC, dialed digits, CPID, and calling party name (Caller ID option).

1b. IAM is routed via home STP of originating SSP.

2. Destination switch (SSP) checks the dialed number against its routing table and confirms
that the called party's line is available for ringing.

2.a Destination SSP transmits ACM to the originating SSP via its home STP to confirm that
the remote end of the trunk circuit has been reserved.

2b. The STP routes the ACM to the originating SSP which connects the calling party's line to
the trunk to complete the voice circuit. The calling party hears ring back tone.

3a. Called party goes off-hook. Destination switch terminates ringing tone and transmits an
ISUP answer message (ANM) to the originating switch via its home STP.

3b. STP routes ANM to originating switch which verifies that the calling party is connected
to the reserved trunk. Billing is initiated.

4a./b. If the calling party hangs up first, the originating switch sends an ISUP release
message (REL) to release the trunk between the two switches. If the called party releases
first, the destination switch sends an REL message to the originating switch to release the
circuit.

5a. When the destination switch receives the REL, it disconnects and idles the trunk and
transmits an ISUP release complete message (RLC) to the originating switch to
acknowledge the release of the remote end of the circuit.

5b. When the originating switch receives or sends an RLC, the billing cycle ends, and the
trunk state is returned to idle.
D. Important Acronyms

1. FXO (Foreign eXchange Office)

• Definition: FXO ports connect to the Public Switched Telephone Network (PSTN).

• Function: They receive the telephone line from the PSTN and can generate off-hook
and on-hook signals.

• Usage: Typically used to connect a gateway or PBX (Private Branch Exchange) to


the PSTN, allowing for outbound and inbound calls.

2. FXS (Foreign eXchange Subscriber)

• Definition: FXS ports connect to end-user devices, such as telephones or fax


machines.

• Function: They provide the necessary signals like dial tone, ringing, and battery to
the connected devices.

• Usage: Commonly found in devices like analog telephone adapters (ATAs), which
allow traditional phones to connect to VoIP networks.

3. SLIC - Subscriber Line Interface Circuit: This is a device that connects the
subscriber's telephone line to the central office, providing the necessary interface
for signaling and power.
4. CODEC - Coder-Decoder: This is a device or software that encodes or decodes a
digital data stream or signal. In the context of DTMF, it converts the analog signals
from the telephone into a digital format for transmission and vice versa.
5. DAC - Digital-to-Analog Converter: This device converts digital data (often from a
computer or digital signal processor) into an analog signal. In telecom, it’s used to
convert the digital DTMF signals back into analog for transmission over traditional
phone lines.
6. IVRS - Interactive Voice Response System: This is a technology that allows a
computer to interact with humans through the use of voice and DTMF tones input
via a keypad. It’s commonly used in customer service applications.
7. CID - Caller ID: This feature allows the recipient of a call to see the caller's phone
number and sometimes the name before answering. It often uses DTMF tones to
transmit the caller's information.
8. NTI (Network Termination Identifier)

• Definition: NTI is a unique identifier used to distinguish a specific network


termination point within a CCS network. It's like a "street address" for a particular
network connection point.

• Purpose: NTI helps in routing signaling messages to the correct destination within
the CCS network.

• Example: Think of a call being routed from one city to another. Each city's switching
system has its own NTI, allowing the CCS system to correctly direct the signaling
information for that call.

9. ET (Equipment Type)

• Definition: ET is a code that identifies the type of equipment used at a particular


network termination point. It helps the CCS system understand the capabilities of
the connected equipment.

• Purpose: ET enables the CCS system to send appropriate signaling messages based
on the type of equipment involved.

• Example: If a network termination point uses a specific type of switch, the ET code
will indicate this, allowing the CCS system to send the correct signaling messages
for that switch.

10. Service Switching Point (SSP)

• Definition: The SSP is a type of telephone exchange that initiates the call
processing when a user dials a number. It is responsible for sending queries to the
network to determine how to route the call.

• Functionality:

o Call Setup: When a call is made, the SSP sends a request to the Service
Control Point (SCP) to retrieve routing information.

o Service Execution: It can also execute various services like call


forwarding, number translation, and more.

• Example: When you dial a number, the SSP is the first point of contact that
processes your call.
11. Signal Transfer Point (STP)

• Definition: The STP acts as a router for signaling messages within the SS7 network. It
routes messages between different nodes, ensuring efficient communication.

• Functionality:

o Message Routing: It directs signaling messages to the appropriate


destination, whether that’s an SSP, SCP, or another STP.

o Load Balancing: STPs can help distribute the signaling load across multiple
paths to optimize performance and reliability.

• Example: Think of the STP as a traffic cop that directs signaling messages to their
correct destinations within the network.

12. OPC: Originating Point Code

This is a unique identifier for a signaling point in the SS7 network. It indicates the source
of the signaling message, helping to route the call correctly.

13. DPC: Destination Point Code

Similar to the OPC, this is a unique identifier for the destination signaling point in the
SS7 network. It indicates where the signaling message is intended to go.

14. CIC: Circuit Identification Code

This code identifies a specific circuit within a trunk group. It helps in managing and
routing calls through the network.

15. CPID: Calling Party Identification

This refers to the information that identifies the calling party, which can include the
caller's number and name, depending on the service provider's capabilities.

In telecommunications and networking, CONN ACK and RELEASE ACK are terms
related to the acknowledgment process during connection establishment and
termination. Let’s break them down:

16. CONN ACK (Connection Acknowledgment)

• Purpose: This message is sent to acknowledge the successful establishment of a


connection between two devices or processes.
• Context: It typically occurs after a connection request (like a SYN packet in TCP) has
been received and accepted. The sender of the CONN ACK confirms that it is ready
to communicate.

• Example: In a TCP connection, after the initial SYN packet is sent, the receiver
responds with a SYN-ACK, and the sender then sends a ACK to finalize the
connection.

17. RELEASE ACK (Release Acknowledgment)

• Purpose: This message is sent to confirm that a connection has been successfully
terminated.

• Context: After a device sends a request to release a connection (like a FIN packet
in TCP), the receiving device sends a RELEASE ACK to acknowledge that it has
received the termination request and has completed the disconnection process.

• Example: In TCP, after the sender sends a FIN packet to close the connection, the
receiver responds with a FIN-ACK, and the sender then sends a RELEASE ACK to
confirm the closure.

18. SLS - Signaling Link Set

• Definition: A collection of signaling links that are used to connect signaling points in
a telecommunications network. It is part of the SS7 (Signaling System No. 7)
architecture, which facilitates the exchange of signaling messages for call setup,
management, and teardown.

19. SLC - Signaling Link Code

• Definition: A code used to identify a specific signaling link within a Signaling Link
Set (SLS). It helps in managing and routing signaling messages in
telecommunications networks, particularly in the context of SS7.

20. SIO - Service Information Octet

• Definition: This refers to a specific octet in signaling messages that carries


information about the service being requested or provided. It is crucial for the
correct interpretation of the signaling messages in various telecommunications
protocols.

21. ANM (Answer Message)

• Definition: The Answer Message (ANM) is a signaling message sent in the backward
direction to indicate that the called party has answered the call.
o Call Confirmation: It confirms that the connection has been established
between the calling and called parties.

o Billing Trigger: The ANM message may be used to trigger billing for the call, as
it signifies that the call is now active.

o When the called party answers the call, the SSP (Service Switching
Point) sends the ANM message back to the calling party's SSP.

o This message verifies that the calling party's line is connected to the reserved
trunk, allowing the call to proceed.

IV. Modern Enhancements and Integration with IP Networks

A. Transition from PSTN to VoIP and IP Networks

• Many telecom providers are replacing PSTN with IP-based networks.

• Softswitches replace traditional circuit switches, using software for call routing.

PTSN Vs VOIP
RJ 45 Cable for VoIP and RJ 11 Cable for Landline Phones

PSTN to VoIP converter


B. Role of PSTN in Contemporary Telecommunications

• PSTN remains crucial for emergency services and reliable communication in areas
with limited internet access.

• Hybrid networks integrating PSTN and VoIP are common.

C. Future of PTSN

1. Transition to Digital: The PSTN is increasingly being replaced by digital


communication technologies. This includes the widespread adoption of Voice over
Internet Protocol (VoIP), which allows calls to be made over the internet rather than
traditional phone lines.

2. PSTN Switch-Off: Many countries are planning to phase out PSTN services entirely.
For instance, the UK has announced that it will completely switch off PSTN services
by 2027. This marks a significant milestone in the modernization of
telecommunications.

3. Enhanced Connectivity: The shift away from PSTN is expected to usher in an era
of enhanced connectivity and advanced communication technologies. This
includes better integration with mobile networks and the use of fiber-optic lines,
which provide faster and more reliable service.
4. Challenges for Legacy Systems: While the PSTN may continue to play a role in
certain areas, especially in rural or less developed regions, companies relying on
legacy systems will need to adapt to newer technologies to remain competitive and
meet customer demands.

5. Future Prospects: The future of PSTN is uncertain, but it is clear that the industry is
moving towards more efficient and flexible communication solutions. Businesses
and consumers alike will benefit from the advancements in technology that come
with this transition.

References

1. What is VoIP

https://www.youtube.com/watch?v=n_b2sJP2pBk&t=287s

2. ISUP and TCAP


https://docstore.mik.ua/univercd/cc/td/doc/product/tel_pswt/vco_prod/ss7_fu
nd/ss7fun05.htm

3. Evolution of PSTN
https://www.ooma.com/blog/business/public-switched-telephone-network-pstn-
evolution/?srsltid=AfmBOopLjAVGl1QLozAHpNHCmtJlqMaP0Sx_wH9sN1KGA2MUN
_P8qYB2&fbclid=IwY2xjawIyr9ZleHRuA2FlbQIxMAABHTy_fO2nuJWejlCTmz_FX_AYfl-
x8f577HvhWyUBiuAsYUdFBKos6VxZMA_aem_Dx893uUx-Far-Q7hGdaTrA

4. How PSTN Works?


https://www.youtube.com/watch?v=6XiEY5n9MGQ&fbclid=IwY2xjawIyr_FleHRuA2Fl
bQIxMAABHbvJ8IX1AU9vLGcIw9-
4xn2EfkmmtpyYgXCLQ6sXSx8OwCgaTtq2WjQsBQ_aem_E7fA4mt1c8a8nuIzb-J1dw

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