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VOICE OVER INTERNET PROTOCOL

SIP tunneling vs PSTN vs FreePBX


vs Asterisk
Nicholas
Dec 21, 2022 5 min

Session Initiation Protocol (SIP) tunneling is a technique used to bypass network


security measures that block or restrict SIP traffic. It involves encapsulating SIP
traffic in a different type of network traffic, such as HTTP, in order to bypass
firewalls or other security measures that are designed to block SIP traffic.

SIP tunneling is often used in situations where SIP traffic is blocked or restricted,
such as in corporate networks or in countries where VoIP services are heavily
regulated. It allows users to make and receive VoIP calls using SIP-based
applications, even if the network they are using does not allow SIP traffic.

There are a number of tools that can be used for SIP tunneling, depending on the
specific needs and constraints of the network. Some common tools include:

SIP TLS tunnel: This is a tool that allows you to encrypt SIP traffic using
Transport Layer Security (TLS) and send it through a TLS-enabled server or
proxy. The server or proxy acts as a "tunnel" that allows the encrypted SIP traffic
to bypass security measures that block or restrict SIP traffic.

SIP over HTTP: This is a tool that allows you to encapsulate SIP traffic in HTTP
packets and send it through a web server or proxy. The web server or proxy acts
as a "tunnel" that allows the SIP traffic to bypass security measures that block SIP
traffic.

SIP over UDP: This is a tool that allows you to encapsulate SIP traffic in User
Datagram Protocol (UDP) packets and send it through a UDP-based server or
proxy. The server or proxy acts as a "tunnel" that allows the SIP traffic to bypass
security measures that block or restrict SIP traffic.

SIP proxy: A SIP proxy is a server that acts as an intermediary for SIP traffic,
routing SIP messages between clients and servers. Some SIP proxies can be
configured to support SIP tunneling, allowing them to act as "tunnels" for SIP
traffic and bypass security measures that block or restrict SIP traffic.

Public Switched Telephone Network (PSTN)


Public Switched Telephone Network (PSTN) is the traditional telephone network
that uses circuit-switched connections to transmit calls over copper wires or fiber-
optic cables. PSTN is the primary means of telephone communication in most
countries around the world, and is used for both local and long-distance calls.

PSTN networks are composed of a series of interconnected telephone exchanges,


which are connected by physical cables or by satellite links. When a user makes a
telephone call, the call is routed through the PSTN network to the destination
telephone exchange, which then connects the call to the intended recipient.

PSTN networks use a variety of technologies and protocols to transmit calls,


including analog signaling and digital signaling protocols such as ISDN (Integrated
Services Digital Network). PSTN networks are also capable of providing additional
services such as voicemail, caller ID, and call forwarding.

PSTN networks have been in use for over a century, and are still the primary means
of telephone communication in many parts of the world. However, with the rise of
internet-based communication technologies such as VoIP (Voice over Internet
Protocol), the use of PSTN networks has been decreasing in some regions.

SIP tunneling can be used to connect SIP-based systems to PSTN networks in a


number of ways, depending on the specific needs and constraints of the network.
Some common techniques include:

Using a gateway: A gateway is a device that allows different types of


communication systems to connect and communicate with each other. A SIP
gateway can be used to connect a SIP-based system to a PSTN network, allowing
users to make and receive calls using SIP-based applications over the PSTN
network.

Using a SIP proxy: A SIP proxy is a server that acts as an intermediary for SIP
traffic, routing SIP messages between clients and servers. Some SIP proxies can
be configured to support SIP tunneling and act as a "tunnel" for SIP traffic,
allowing it to bypass security measures that block or restrict SIP traffic and
connect to PSTN networks.

FreePBX
FreePBX is an open-source graphical user interface (GUI) for managing Asterisk, a
free and open-source private branch exchange (PBX) software. FreePBX allows users
to easily set up and manage Asterisk-based PBX systems, including features such as
voicemail, call routing, and call waiting.

FreePBX can be used in conjunction with Public Switched Telephone Network


(PSTN) networks to allow users to make and receive telephone calls using traditional
telephone lines. To do this, a PSTN gateway or interface is typically used to connect
the FreePBX system to the PSTN network. The PSTN gateway or interface acts as a
bridge between the FreePBX system and the PSTN network, allowing users to make
and receive calls using their FreePBX system over the PSTN network.

FreePBX can also be used with VoIP (Voice over Internet Protocol) networks to allow
users to make and receive telephone calls over the Internet. In this case, a VoIP
service provider or SIP (Session Initiation Protocol) trunk is typically used to
connect the FreePBX system to the VoIP network. The VoIP service provider or SIP
trunk acts as a bridge between the FreePBX system and the VoIP network, allowing
users to make and receive calls using their FreePBX system over the Internet.

Asterisk
Asterisk is a free and open-source private branch exchange (PBX) software that
allows users to set up and manage telephone systems. It is developed and maintained
by the Asterisk Project, a community-driven effort to create a PBX software that is
flexible, feature-rich, and easy to use.

Asterisk is often used as the foundation for PBX systems, providing the underlying
infrastructure and features that allow users to make and receive telephone calls, as
well as access a variety of additional features such as voicemail, call routing, call
waiting, and caller ID. It is highly customizable and can be configured to meet the
specific needs of different organizations and users.

Asterisk supports a wide range of communication protocols, including SIP (Session


Initiation Protocol), IAX (Inter-Asterisk Exchange), H.323, and MGCP (Media Gateway
Control Protocol), as well as traditional PSTN (Public Switched Telephone Network)
protocols such as ISDN (Integrated Services Digital Network) and SS7 (Signaling
System 7). This allows it to connect to a variety of different telephone systems and
networks, including VoIP (Voice over Internet Protocol) networks, PSTN networks,
and hybrid systems that combine both types of networks.

Asterisk is used by a wide range of organizations, from small businesses to large


enterprises, and is known for its flexibility, reliability, and scalability. It is often used
in conjunction with graphical user interfaces (GUIs) such as FreePBX, which provide
an easy-to-use interface for managing Asterisk-based PBX systems.

FreePBX vs Asterisk
FreePBX is a graphical user interface (GUI) for managing Asterisk, a free and open-
source private branch exchange (PBX) software. FreePBX allows users to easily set up
and manage Asterisk-based PBX systems, including features such as voicemail, call
routing, and call waiting.

Asterisk is the underlying PBX software that powers FreePBX. It provides the
infrastructure and features that allow users to make and receive telephone calls, as
well as access a variety of additional features such as voicemail, call routing, call
waiting, and caller ID. Asterisk is highly customizable and can be configured to meet
the specific needs of different organizations and users.

In other words, FreePBX is a front-end tool that sits on top of Asterisk and provides
an easy-to-use interface for managing Asterisk-based PBX systems. Asterisk is the
underlying software that provides the PBX functionality and is responsible for
handling the actual communication between telephones.

FreePBX and Asterisk can be used together to create a powerful and flexible PBX
system that allows users to make and receive telephone calls, as well as access a
variety of additional features. They are often used in conjunction with traditional
PSTN (Public Switched Telephone Network) networks, as well as VoIP (Voice over
Internet Protocol) networks, to allow users to make and receive calls using a variety
of different communication technologies.

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