Showing posts with label Carrier. Show all posts
Showing posts with label Carrier. Show all posts

Friday, January 24, 2014

List of ICX Service Providers in Bangladesh


Bangladesh Telecommunication Regulatory Commission (BTRC) is the official authority of the Bangladesh Government for issuing license of Interconnection Exchange (ICX) Services in the country. Below is the list of all the licensed / registered ICX providers with contact details. List is prepared based on the information’s provided on BTRC website. Please visit for more details.

Last Updated: August 2019

Agni Systems Limited
Address: Navana Tower (11th Floor), Suite-A, 45 Gulshan Avenue, Gulshan-1, Dhaka-1212
Date of Issue: 12-04-2012

Bangladesh Telecommunications Company Limited - BTCL
Address: Telejogajog Bhaban, 37/E, Eskaton Garden, Dhaka-1000
Tel: 02-8311500, Fax: 02-8312577
Web: www.btcl.gov.bd
Date of Issue: 20-03-2008

Bangla ICX Ltd.
Address: Plot#68, (Old#45), Block-H, Road#11, Banani, Dhaka-1213
Date of Issue: 12-04-2012

Bangla Telecom Limited
Address: Level-4, House-3, Road-7, Block-F, Banani, Dhaka- 1213
1Date of Issue: 2-04-2012

Bantel Limited
Address: Sumona Goni Trade Center, Plot-2, Level-2 & 6, Panthapath, Kawran Bazar, Dhaka-1215
Date of Issue: 12-04-2012

Cloud Tel Limited
Address: Wasa Bhaban, 3rd Floor, 98 Kazi Nazrul Islam Avenue, Kawran Bazar, Dhaka-1215
Date of Issue: 12-04-2012

Cross World Telecom Ltd.
Address: UCEP Cheyne Tower (1st Floor), 25, Segun Bagicha, Dhaka-1000
Date of Issue: 12-04-2012

Gazi Networks Limited
Address: 37/2 Purana Paltan (3rd Floor), Dhaka
Date of Issue: 12-04-2012

Getco Telecommunications Limited
Address: 26 Shyamoli, Mirpur Road, Dhaka-1207
Tel: 02-8158718-21, Fax: 028113319
E-mail: info@getco.com.bd
Web: www.getco.com.bd
Date of Issue: 25-02-2008

Imam Network Ltd.
Address: Amin Court (2nd Floor), 62-63, Motijheel C/A, Dhaka-1000
Date of Issue: 12-04-2012

Integrated Services Limited
Address: House no: 55, Road: 4/A, Dhanmondi R/A, Dhaka- 1209
Date of Issue: 12-04-2012

Jibondhara Solutions Ltd.
Address: Level-05, UTC Building, 8 Panthapath, Dhaka-1215
Date of Issue: 12-04-2012

M & H Telecom
Address: Bay Tower (3rd Floor), 50 Mohakhali C/A, Dhaka-1212
Tel: 02-9886913, 9887181, 9887168, 9889806
Mobile: 01713-033826, Fax: 02-9887552
E-mail: info@mnhtel.com
Web: www.mnhtel.com
Date of Issue: 25-02-2008

M. M. Communications Ltd.
Address: 11F, Sotabdi Tower, 292 Fokirapul Inner Circular Road, Dhaka-1000
Date of Issue: 12-04-2012

Micro Trade
Address: Aqua Tower (8th Floor), 43 Mohakhali C/A, Dhaka-1212
Date of Issue: 12-04-2012

Mother Telecommunication
Address: Safura Tower, Level-10, 20Kemal Ataturk Avenue, Banani, Dhaka-1213
Date of Issue: 12-04-2012

New Generation Telecom Limited
Address: Land Mark Building, 10th Floor, 12-14 Gulshan North C/A, Dhaka-1212
Date of Issue: 12-04-2012

Paradise Telecom Ltd.
Address: Awal Centre, 34, Kemal Ataturk Avenue (12th Floor), Banani C/A, Dhaka-1213
Date of Issue: 12-04-2012

Purple Telecom Limited
Address: Rupsha Tower, Suite-B2, Plot-7, R#17, Banani C/A, Dhaka-1212
Date of Issue: 12-04-2012

Ring Tech (Bangladesh) Ltd.
Address: 19/1, North Dhanmondi, Kalabagan, Dhaka-1205
Date of Issue: 12-04-2012

Softex Communication Limited
Address: Suite-B, Level-6, Concord I-K Tower, Plot-2, Block- Cen(A), North Avenue, Gulshan-2, Dhaka-1212
Date of Issue: 12-04-2012

S R Telecom Limited
Address: 104, 6th Floor, Delwara Bhaban, Agrabad C/A, Chittagong
Date of Issue: 12-04-2012

Teleplus Newyork Limited
Address: Ismail mansion (3rd Floor), 32 Mohakhali C/A, Dhaka- 1212
Date of Issue: 12-04-2012

Tele Exchange Limited
Address: Green Delta Aims Tower (11th Floor), 51-52, Mohakhali C/A, Dhaka-1213
Date of Issue: 12-04-2012

Vertex Communications Ltd.
Address: 18, Karwan Bazar C/A, Dhaka-1215
Date of Issue: 03-05-2012

VOICETEL LTD.
Address: Shanta Western Tower, Level-13, Bir Uttam Mir Shawkat Road, 186, Tejgaon I/A, Dhaka-1208
Date of Issue: 12-04-2012

For update list of Interconnection Exchange (ICX) Service Providers in Bangladesh and further details please visit the BTRC web portal.


Related:
List of Licensed Call Centers in Bangladesh
BPO | Business Process Outsourcing List in BD
IGW Operator License List in Bangladesh
ISP | Internet Service Provider in BD | Central Zone 

Friday, January 10, 2014

List of VSAT Providers and Users in Bangladesh



Bangladesh Telecommunication Regulatory Commission (BTRC) was formed on 31st January of 2002 under the Bangladesh Telecommunication Regulatory Act 2001. BTRC is the authority of the Bangladesh Government to provide and facilitate license for some sort of Internet and Telecommunication related services or devices across the country. As per the information provided on the official website of BTRC, below is the list of all the VSAT providers and users in the country. Please visit the website for updated lists and more details.

List of Registered VSAT Provider of Bangladesh:

PCCW Global (HK) Limited
Registered Address: PCCW Tower (33rd floor), Taikoo Place, Quarry Bay, Hong Kong
Local Agent: M/s. Dactel Systems
Address: House# B101 (3rd floor), Road#7, New DOHS, Dhaka- 1206, Bangladesh
Contact: 02-8754667, Fax: 02-8754665
E-mail: info@dactelsystems.com
License No: BTRC/LL/VSAT-P(17)PCCW(Hong Kong)/2008-26, Dated: 24-02-2010

Bangladesh Export Import Company Limited
Registered Address: House# 17, Road # 2, Dhanmondi R/A, Dhaka-1205
Tel: 02-8815559, Fax: 02-8815747
E-mail: info@bol-online.com
License No: BTRC/LLIVSAT-P(18)Beximco(Dhaka)/2009-21, Dated: 11-11-2009


List of Registered VSAT Provider with Hub of Bangladesh:

DNS SatComm Limited
Registered Address: House# 26, Road# 28, Block# K, Banani, Dhaka- 1213, Bangladesh
Contact: 02-9888712, 8817270, 9893308-9, Fax: 02-8815490
E-mail: info@dnsgroup.com.bd
Web: www.vsat.com.bd
Satellite Earth Station, NOC, Monitoring Centre: Plot# 278, Export Processing Zone (EPZ), Savar, Dhaka
Contact: 02-7702721
License No: BTRC/LL/VSAT-P/H(10)DNSSatComm/2007-6, Dated: 30-07-2009

Square Informatix Limited
Registered Address: Square Center (Level-11), 48 Mohakhali C/A, Dhaka-1212, Bangladesh
Contact: 02-8833047-56, 8859007, Fax: 02-8828768, 8828609
E-mail: sil@squaregroup.com
Web: www.e-home2u.com
License No: BTRC/LL/VSAT-P/H(2)Square(Dhaka)/2007-8, Dated: 27-07-2010

ADN Telecom Limited
Registered Address: Red Crescent Concord Tower (19th Floor), 17 Mohakhali C/A, Dhaka-1212, Bangladesh
Tel: 02-8861234-8, Fax: 02-9887551
E-mail: irshad@atcl.net
License No: BTRC/LL/VSAT-P/H(58)ADNSL(Dhaka)/2006-9, Dated: 28-02-2007


List of Registered VSAT Users in Bangladesh:

United Nations High Commissioner for Refugees
Address: UNHCR Sub Office, New Cottages, Motel Road, Cox's Bazar-4700
Tel: 0341-63550, Fax: 0341-63870
E-mail: bgdco@unhcr.org
License No: BTRC/LL/VSAT-U(122)UNITEDNATIONS(Cox'sBazar)/2006-104, Dated: 07-09-2006

Bangladesh Meteorological Department
Address: Abhawa Bhaban, Agargaon, Dhaka
License NO: BTRC/LL/VSAT-U(128)Meteorological(Dhaka)/2006-19, Dated: 15-06-2008

Bangladesh Meteorological Department
Address: Abhawa Bhaban, Agargaon, Dhaka
License NO: BTRC/LL/VSAT-U(126)Meteorological(Cox’sBazar)/2006-20, Dated: 15-06-2008

Bangladesh Meteorological Department
Address: Abhawa Bhaban, Agargaon, Dhaka
License NO: BTRC/LL/VSAT-U(13)Meteorological(Cox’s Bazar)/2007-21, Dated: 15-06-2008

Bangladesh Meteorological Department
Address: Abhawa Bhaban, Agargaon, Dhaka
License NO: BTRC/LL/VSAT-U(65)Meteorological(Moulvibazar)/2008-48, Dated: 26-05-2009

Bangladesh Meteorological Department
Address: Abhawa Bhaban, Agargaon, Dhaka
Contact: 02-8116634
Email: info@bmd.gov.bd
License NO: BTRC/LL/VSAT-U(135)Meteorological(Rangpur)/2012-923, Dated: 02-08-2012

Bangladesh Meteorological Department
Address: Abhawa Bhaban, Agargaon, Dhaka
Contact: 02-8116634
Email: info@bmd.gov.bd
License NO: BTRC/LL/VSATU(136)Meteorological(Dhaka)/2012-924, Dated: 02-08-2012

Comprehensive Nuclear-Test-Ban Treaty Organization (CTBTO)
Address: Vienna International Center, P.O Box-1200, A-1400, Vienna, Australia.
Local Agent: Bangladesh Atomic Energy Commission (BAEC)
Address: 4 Kazi Nazrul Islam Avenue, GPO Box: 158, Ramna, Dhaka, Bangladesh.
License No: BTRC/LL/VSATU(38)CTBTO(Dhaka)/2008-66, Dated: 07-07-2010

Comprehensive Nuclear-Test-Ban Treaty Organization (CTBTO)
Address: Vienna International Center, P.O Box-1200, A-1400, Vienna, Austria.
Local Agent: Bangladesh Atomic Energy Commission (BAEC)
Address: 4 Kazi Nazrul Islam Avenue, GPO Box: 158, Ramna, Dhaka, Bangladesh.
License No: BTRC/LL/VSATU(57)CTBTO(Chittagong)/2008-67, Dated: 07-07-2010

United Nations Children's Fund (UNICEF)
Address: UNICEF Bangladesh Country Office, GPO Box: 58, BSL Office Complex (3rd Floor), 1 Minto Road, Dhaka-1000
License No: BTRC/LL/VSATU(37)Unicef(Dhaka)/2008-69, Dated: 04-07-2010

The Embassy of Japan
Address: Plot # 5 & 7, Dutabash Road, Baridhara, Dhaka-1212
License No: BTRC/LL/VSAT-U(31)Japan Embassy(Dhaka)/2007-70, Dated: 27-07-2010

Civil Aviation Authority of Bangladesh
Address: Civil Aviation Head Office, Kurmitola, Dhaka
Contact: 02-7912366
Email: chairman@caab.gov.bd
License: BTRC/LL/VSAT-U(137)CAAB(Dhaka)2012-45, Dated: 16-01-2013


source: BTRC
List updated: August 2019


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Friday, September 30, 2011

Wholesale VoIP Solution

The birth of Voice-over-IP technology in late 1990s and its rapid growth in early 2000s, have resulted in the proliferation of many alternative telecom providers. Those providers have introduced a number of innovative communications services such as calling cards and callback which relied on call origination and termination over the Internet.

The growing number of new telecom providers led to increased demand for traffic aggregation services and as a result the Wholesale VoIP business was born. Wholesale VoIP providers act as traffic aggregates and/or traffic exchanges for their customers.

They sign up contracts with both retail and wholesale VoIP providers and act as a middle man for call origination and termination services. When partners send traffic to the Wholesale VoIP provider, he/she reroutes it to other partners for termination and makes profit from the difference in negotiated rates.



Business Solution
 
PROVIDER Softswitch will offer end-to-end, cost-effective and scalable Wholesale VoIP solution. That solution will feature powerful billing, flexible routing, and proven interoperability with equipment from other leading VoIP vendors.

Because all solution components will be developed by PROVIDER and Softswitch manufacturer we can eliminate integration issues and for that we can quickly start and/or expand our Wholesale VoIP business while enjoying high return on investment.


How will the solution work?

PROVIDER Experience
  • PROVIDER will receive VOIP Traffic request from Partner 1 for termination.
  • Then PROVIDER Softswitch will accept the traffic and will send authorization request to PROVIDER VoIP Billing server. The VoIP billing server will verify the account balance of Partner 1 and will authorize call termination.
  • PROVIDER Softswitch will re-route the traffic to Partner 2, hiding traffic source information.
  • Upon completing the call, the PROVIDER VoIP Billing server records will debit the account of Partner 1 and will record CDR record of the call.


Solution components are required
 
Core elements of PROVIDER Wholesale VoIP Solution include:
  • PROVIDER VOIP Billing Server is a robust billing server that provides our customers with all necessary tools to successfully implement a wide spectrum of VoIP business models.
  • PROVIDER Softswitch is an advanced VoIP softswitch that provides our customer with secure and reliable peering between their own networks and the VoIP networks of their business partners.
Other Posts:
CallShop Solution

Tuesday, August 16, 2011

PSTN vs VoIP


VoIP (Voice over Internet Protocol) is going to be more popular around the world comparatively with PSTN (Public Switched Telephone Network). The main reason for growing market for VoIP over PSTN is availability, reliability and low cost etc. In this article I will show that there is no reason to limit the expectations to achieve only the same level of quality as in PSTN.

By deploying wideband codec, much better voice quality can be achieved. But using narrowband codec there are some standard solution that can be used in VoIP to get finest quality voice. Such as, traditional PSTN solution does not use the available spectral bandwidth, but these can be done easily in VoIP system. Implementing a wideband codec or expanding the bandwidth of narrowband codec does not automatically guarantee great quality. There are many potential pitfalls when deploying VoIP. There are some issues need to be discussed that effect the final voice quality over PSTN or VoIP. In this article I will discuss what level of quality can be achieved and describe how this can be implemented.



Speech Signals and Speech Coding Sampled digital signals can hold frequency content up to half the sampling frequency. As we know, a young adult has a hearing span from about 20 Hz to 20,000 Hz.
And the sampling frequency of CD audio is chosen to be 44.1 kHz, which is more than double that of the highest frequency perceivable by most humans. Legacy telephony solutions are narrowband, which seriously limits the achievable quality. Wideband codec could potentially be used in digital telephone systems, but this has never been practical enough to gain any real interest

In traditional telephony applications such as PSTN, the speech bandwidth is restricted much more than the inherent limitations of narrowband coding. Typical telephony speech is band limited to 300 Hz to 3400 Hz (listen to Sound Pure digital connections are found in enterprise environments.
Due to poor connections or old wires, significant distortion is often generated in the analog part of the phone connection, a type of distortion that is entirely absent from VoIP implementations. The cordless phones so popular today also generate significant amounts of analog distortion due to radio interference and other implementation issues.



Calls between two parties in a PSTN (Public Switched Telephone Network) are connect by a series of private or public switches. The resulting fixed communications link is dedicated for the duration of the call. When an individual makes a phone call over a circuit-switched network, a connection is made between the providers PBX and the local telephone company, also known as the PSTN. Depending on the destination, this connection might extend to the national or international exchange before reaching another local exchange, where it will be passed on to the PBX and the person who receives the call. This end-to-end link, established by a series of public and private switches, is 100% dedicated on a single, per-call basis and cannot be shared or used for another function as long as the call is in progress. For this reason, these dedicated circuits cannot be shared and the carrier bills the call on a time and distance rate.

But the Internet does not use switches to link calling parties. Instead, the analog voice signal is digitized by an Internet Protocol (IP) and broken up into thousands of small data packets by a router – the VoIP equivalent to a switch. These data packets are sent, or routed, over the public Internet to their destination, enabling calls to bypass the PSTN entirely.


Other article can be read:

Thursday, August 11, 2011

What is H.323 Signaling Protocol - A basic


To provide some special services through the internet (IP based network) such as real-time audio / video communication, data communication H.323 is a standard signaling protocol which components, protocols and procedures. H.323 is part of a family of ITU-T recommendations called H.32x that provides multimedia communication services over a variety of networks.

H.323 was invented to provide various multimedia applications through LAN (Local Area Network). But due to its popularity and rapidly growing services it has evolved in needs of VoIP (Voice over Internet Protocol). One important strength of H.323 was the relatively early availability of a set of standards. It was not only used to define basic call models but in addition to supplementary services that can easily involved in Business Communications. The first VoIP standard protocol is H.323 to adopt IETF standard RTP to transport audio and video communications over the internet (IP network). H.323 standard specifies four kinds of components, when networked together, provide the "point-to-point" and "point-to-multipoint" multimedia-communication services. These are:
  • Terminals
  • Gateway
  • Gatekeepers
  • Multipoint Control Unit

Terminals: Used for real-time bi-directional multimedia communications, an H.323 terminal can either be a personal computer (PC) or a stand-alone device, running an H.323 and the multimedia applications. It supports audio communications and can optionally support video or data communications. Because the basic service provided by an H.323 terminal is audio communications, an H.323 terminal plays a key role in IP–telephony services. The primary goal of H.323 is to inter-work with other multimedia terminals. H.323 terminals are compatible with H.324 terminals on SCN and wireless networks, H.310 terminals on B– ISDN, H.320 terminals on ISDN, H.321 terminals on B–ISDN, and H.322 terminals on guaranteed QoS LANs. H.323 terminals may be used in multipoint conferences.

Gateways: A gateway connects two dissimilar networks. An H.323 gateway provides connectivity between an H.323 network and a non – H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and SCN networks ( SCN networks include all switched telephony networks, e.g., public switched telephone network [PSTN] ). This connectivity of dissimilar networks is achieved by translating protocols for call setup and release, converting media formats between different networks, and transferring information between the networks connected by the gateway. A gateway is not required, however, for communication between two terminals on an H.323 network.

Gatekeepers: A gatekeeper can be considered the brain of the H.323 network. It is the focal point for all calls within the H.323 network. Although they are not required, gatekeepers provide important services such as addressing, authorization and authentication of terminals and gateways; bandwidth management; accounting; billing; and charging. Gatekeepers may also provide call-routing services.

Multipoint Control Units: MCUs provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video coder/decoder (CODEC) to use, and may handle the media stream. The gatekeepers, gateways, and MCUs are logically separate components of the H.323 standard but can be implemented as a single physical device.


Key Benefits of H.323:

Codec Standards: 
H.323 establishes standards for compression and decompression of audio and video data streams, ensuring that equipment from different vendors will have some area of common support.

Interoperability:
Users want to conference without worrying about compatibility at the receiving point. Besides ensuring that the receiver can decompress the information, H.323 establishes methods for receiving clients to communicate capabilities to the sender. The standard also establishes common call setup and control protocols.

Network Independence:
H.323 is designed to run on top of common network architectures. As network technology evolves, and as bandwidth management techniques improve, H.323-based solutions will be able to take advantage of those enhanced capabilities.

Platform and Application Independence:
H.323 is not tied to any hardware or operating system. H.323-compliant platforms will be available in many sizes and shapes, including video-enabled personal computers, dedicated platforms, IP-enabled telephone handsets, cable TV set-top boxes and turnkey boxes.

Bandwidth Management:
Video and audio traffic is bandwidth-intensive and could clog the corporate network. H.323 addresses this issue by providing bandwidth management. Network managers can limit the number of simultaneous H.323 connections within their network or the amount of bandwidth available to H.323 applications. These limits ensure that critical traffic will not be disrupted.

Flexibility:
An H.323 conference can include endpoints with different capabilities. For example, a terminal with audio-only capabilities can participate in a conference with terminals that have video and/or data capabilities. Furthermore, an H.323 multimedia terminal can share the data portion of a video conference with a T.120 data-only terminal, while sharing voice, video, and data with other H.323 terminals.

Inter-Network Conferencing:
Many users want to conference from a LAN to a remote site. For example, H.323 establishes a means of linking LAN-based desktop systems with ISDN-based group systems. H.323 uses common codec technology from different videoconferencing standards to minimize transcoding delays and to provide optimum performance.

Related Other Topics:

Monday, July 4, 2011

JAJAH Mobile Web - Direct Call from Mobile Web Browser


Make calls directly from your web-enabled mobile handset and save money when talking. Using the browser of your web-enabled Smart phone (Blackberry, Treo, iPAD, Smart phones by Motorola, Nokia, SonyEricsson, Samsung etc..) you can make one-click free global call around the world. No download is required to use JAJAH Mobile Web. User can make calls directly by log in with their username and password.

Requirements:


  • Run on special phone which runs on Microsoft Windows Mobile, Microsoft Windows CE, Pocket PC, Symbian Operating System.
  • JAJAH Mobile Web service is compatible with any standard mobile handset with internet connection that has WAP 2.0 or XHTML web browser.
  • It works any mobile network in any countries that has GSM and CDMA network which includes 99% of all with a minimum of 2.0 G dada service.
  • For GSM Network GPRS or EDGE is required and for CDMA network 1X(RTT) and EVDO is required.
  • For both type of networks use the term 3G and HSPDA to talk about high speed mobile data networks. All of this network type allows the end-user to use JAJAH mobile web.

Features:
  • User can make free calls to other JAJAH users and to make calls all over the world (any Landline or Mobile number) the JAJAH users can make calls with a competitive and low price. JAJAH mobile web is different if compare with other VoIP service provider because some of its special features. User can directly make calls all over the world simply using their mobile web browser. For details rate plan Click here.

  • In fall 2006 JAJAH launched a special JAJAH Mobile plug-in, a downloadable application for mobile phones, it attracted thousands of customers across the globe. Using these service users was able to make calls by sending a SMS. But today the new JAJAH Mobile Web is specially designed fro Smart Phones give opportunity to its users to make low-cost phone calls simply from their mobile web browser.

  • JAJAH Mobile web service can be use when you are in roaming. Roaming is very high-cost service any provider. But JAJAH gives you the opportunity to save this cost. TO do so you need to change your registered mobile number. During travel you can buy a local SIM card to use in your handset. Now using JAJAH mobile web you can update your registered mobile number by simply pressing the “Change” link at the top of the page. Now in the new page change your registered mobile number entering in the text box. Alternatively, you can choose one of your home, office or mobile numbers as your source number. Finally press the "Save" button and you'll return to the previous page with your updated source number. This system helps you to save your roaming fees while you are in travel.



How to make calls using JAJAH Mobile Web:
  • Open your mobile web browser.
  • Log into your JAJAH account using your username and password if you are already JAJAH user. If you are a new user than you need to get registered with JAJAH service, sign up is absolutely free.
  • For faster access next time you can bookmark this page.
  • Type the number you want to make call or you can select your saved number from your address book. To make calls check your current balance at the top of the page.
  • Now press the “Call” button and start talking.
JAJAH.Mobile Web connects you.


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Tuesday, January 25, 2011

PSTN, Voice and data


Voice and Data:
Telephony is the communication of spoken information between two or more participants or devices, by means of carried signals over wires or wireless. Ever since Alexander Graham Bell invented the telephone circuit and first envisioned the public telephone system (PSTN). In today world Consumers and businesses have relied on telephony as a clip of interaction.
The PSTN

In the year 2004, there were almost 200 million traditional, land phone lines in the United States of America. This network connects all these phones together wraps around the world. The main function of the PSTN (Public Switched Telephone Network) is to reliably facilitate telephone conversations at any time of day. The PSTN combines Digital, Analog and electro-mechanical data links that strive to make sure that every time you pick up your phone receiver, you hear a dial-tone, and every time you can make call directly to your friend and family or anyone across the globe whenever you need.
Some key features of PSTN, Voice, PBX and Data

  • The PSTN was developed as a result of premature telephony systems' efforts at building meshed and switched networks.
  • SS7 is the switch-to-switch signaling network using to connect and bill calls placed across the PSTN.
  • POTS (plain old telephone service), is the basic single-line analog voice service from a phone company.
  • Key systems and PBXs provide businesses a method of communicating to others for their business or any other purpose at a lower cost than dealing only with PSTN-provided services.
  • Lines and trunks are links between telephony devices. A line is a link from a switch to a phone. A trunk is a link from a switch to a switch.
  • Voice over IP encompasses a large family of interface technologies, protocols, and standards that enable real-time media applications using IP networks.
  • Traditional telephony isn't scalable like VoIP because it isn't software-driven like VoIP. Its ratio of calling capacity to network infrastructure is fixed, while VoIP's has lots of room for optimization.
  • VoIP is suitable for deployment in homes and businesses, and, thanks to broadband and deregulation, is earning a reputation as a more practical way of delivering telephony services.

With the help of Internet technologies and high-speed data connectivity in the enterprise, a new family of telephony technologies has begun called VoIP. Voice over Internet Protocol, or VoIP, has significant appeal for the enterprise, for service providers, and for end users, because it allows the Internet and commonplace data networks, like those at offices, factories, and campuses, to become carriers for voice calls, video conferencing, and other real-time media applications.

VoIP-savvy organizations are discovering that they can apply the paradigm of distributed, software-based networking to voice applications and enable a new generation of telecommunications features, cost-savings, and productivity enhancements.

VoIP can replace business telephone systems, or it can add value to existing traditional telephony devices. For instance, long-distance connectivity between two offices with traditional telephone systems can often be accomplished with a lower cost per call when VoIP is employed.

VoIP network protocols can serve as a platform for other communication media like text messaging and video conferencing. In fact, you've probably used a flavor of VoIP for such an application by now; they've been popular as an Internet pastime for several years. Yahoo! offers a "party line"-style service that features Voice over IP chat rooms. Apple's iChat and Microsoft's NetMeeting, Google's gtalk applications also offer text, voice, and video calling delivered through VoIP protocols.

Dozens of standards define how Voice over IP works, but little documentation exists on best practices for implementing and maintaining the technology in the enterprise. There's not much introductory instruction for VoIP, so beginners may have a hard time taking their first steps with it. There have been several high-profile implementation failures among large enterprise adopters, and this may be why IP telephony has such an intimidating reputation.

Nonetheless, if it's done right, Voice over IP can transform the cost model of telecommunications by combining the overhead of voice and data expertise and infrastructure. It can also enhance productivity for end users by introducing new features and for telecommunication administrators by centralizing management functions. Voice over IP can decrease the expense of future computer-telephony integration projects, while making it easier to link voice systems with web servers and database applications.

The book Switching to VoIP by Theodore Wallingford will give you practical guidance on switching to Voice over IP from traditional telephony systems. It gives a brief introduction to traditional telecom systems, and correlates their features and fundamentals to those of IP telephony systems, while showing ways of integrating traditional telephony assets into an IP-based voice network. It will describe the standards involved, so you can make educated choices among the large selection of components and vendors. It will also help you conquer some of the most problematic issues that people face when building telephony systems with Voice over IP.

Plain Old Telephone Service
When you pick up a traditional telephone, like the one in your house, the dial-tone you hear is a signal transmitted over the simplest kind of voice circuit: an analog phone line, also called a POTS (plain old telephone service) line. The line is a simple electrical loop. On the other end of the line is the telephone company's switch, or exchange. It is this switch that gives your phone its electrical power, sends the sound signals emitted by its audio transducer, and knows how to handle the numbers you dial on its keypad.

The switch also knows how to provide POTS calling features like call-waiting, a service that allows your call in progress to be placed on hold while you answer another caller who is trying to call your linewhen you tap the receiver's hook, the switch recognizes the electromagnetic signal you created (called a flash), connects your phone line to the call that's waiting on another line on the switch, and places your first caller on hold. Later, when you send the flash signal again, the first caller is re-connected with your line and you can resume the original conversation. On the PSTN, calling features are always provided by the telephone company switchnot by the phone itself.

Despite historically increasing demand for features and gains in efficiency, POTS has been the most popular type of service provided by phone companies for the last several decades. Its core of a copper loop with an electromagnetic telephone receiver on one end and a central office switch on the other end hasn't changed much during that time. POTS' stalwart reputation and simplicity have been key reasons for its longevity.

So why not keep that copper telephone loop forever? POTS works great in low-density scenarios; that is, places with a small number of simultaneous callers, such as a one-family home. POTS can also excel where extensive calling features aren't important or where the telephone is rarely used. But when density and features become more critical, POTS isn't always the best solution.

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