Modern Telephone Networks and Their Types
Telecommunication technologies have always revolutionized the way humans interact with each other. Since the invention of the first phone by Alexander Graham Bell, we have seen the continuous evolution of technologies that have impacted humanity so much. The speed of change of adaptation of new technology is getting superfast. Let’s have a brief look at the generations of telecom standards or connectivity types available in modern world.
PSTN or Public Switched Telephone Network is simply known as a ‘telephone line’. This is the most commonly used method for all users that only have the need to use one line for one conversation at a time using only one phone number. This is the basic service that you have at home and in a small business. PSTN is also used to connect to internet by converting a line into ADSL or Asymmetric Digital Subscriber line. This type of service is most commonly used by small businesses because it provides enough bandwidth for a small group of users to access the internet. It works only over an existing PSTN, so you need to have an active PSTN to be able to have ADSL.
Integrated Services Digital Network (ISDN) is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the traditional circuits of the public switched telephone network
ISDN is a circuit-switched telephone network system, which also provides access to a packet switched networks, designed to allow digital transmission of voice and data over ordinary telephone copper wires, resulting in potentially better voice quality than an analog phone can provide. An ISDN line is a digital line - your computer connects to the ISDN line (via a terminal adaptor) without having to convert the data into sound first.
ISDN provides two types of access, called basic access and primary access, respectively:
Basic Rate access provides two 'channels', each of which can be used for separate calls, thus equivalent to having two regular lines on a single twisted pair. Two 64 kbit/s channels can be used as independent phone lines, referred to as B-channels, and one 16 kbit/s channel (optional), called D-channel, used for signaling and for the transmission of packet data. Therefore, the basic access, characterized by a total speed of 144 kbit/s in each direction of transmission, is also known as 2B + D access.
Primary Rate Access (PRI): scheduled mainly for business users, offers 31 channels of which thirty 64 kbit/s, with the addition of one D-channel at 64 kbit/s used only for signaling between the user and the network. The primary access, also called 30 B+D access, is realized with a conventional 2 Mbit/s line. PRI is the most popular choice of today’s businesses since the cost of terminating one PRI line is lower than thirty analog trunk lines.
It features rich with facilities such as :
Direct Inward Dialling: For each PRI line, the service provider would provide more around 100-500 numbers which can be used by outsiders to call the extension directly, instead of having to go through the PBX Auto-attendant.
Caller ID: Since all the extensions have their own number, this unique number will be displayed on the phones that they are calling to. Some call center applications are based on the unique caller ID number for differentiation of services.
VoIP is short for Voice over Internet Protocol. Voice over Internet Protocol is a category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls by sending voice data in packets over a LAN or WAN using IP rather than by traditional circuit transmissions of the PSTN. One advantage of VoIP is that the telephone calls over the Internet do not incur a surcharge beyond what the user is paying for Internet access, much in the same way that the user doesn't pay for sending individual emails over the Internet.
Business VoIP is growing in use worldwide. While the technology has countless advantages, it also has its share of disadvantages. Advantages include the huge potential savings and greater scalability of systems. One disadvantage is that it requires a reliable internet connection with high bandwidth availability.
The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, as well as in instant messaging over Internet Protocol (IP) networks. A SIP trunk is a direct connection between your organization and an Internet telephony service provider (ITSP). It enables you to extend voice over IP (VoIP) telephony beyond your organization's firewall without the need for an IP-PSTN gateway. SIP trunking shares similar parallels to VoIP, where the equivalent in the legacy world would be PBX trunks. SIP trunks differ from PBX trunks in that they carry all forms of media, not just voice. Being designed for use in a data network, SIP trunks transmit packets, which could carry voice, data or video.
How to make Unified Communication work for you?
In order to connect a traditional analog phone line to a VoIP system you would need an FXO gateway (or an analog telephone adapter). This allows you to connect your incoming analog FXS line, the line from the telephone provider, to the FXO port of the gateway. The FXS line is the port in your analog phone and the FXO port is the port found in your gateway or ATA. When both ports are connected, the end result is the easy-conversion of analog technology to a much more powerful and feature-rich network: VoIP.
This leads us to create a VoIP network that unifies communications. The term Unified Communications refers to placing all communications under the same umbrella. The benefits of unifying your business communications are endless, but at the end of the day, it all comes down to streamlining. Unifying, managing, integrating and enhancing your business communication into one powerful IP solution will increase productivity all around.