Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. A voice frequency (VF) or voice band is one of the frequencies, within part of the audio range, that is used for the transmission of speech. In telephony, the usable voice frequency band ranges from approximately 300 Hz to 3400 Hz. It is for this reason that the ultra low frequency band of the electromagnetic spectrum between 300 and 3000 Hz is also referred to as voice frequency (despite the fact that this is electromagnetic energy, not acoustic energy). The bandwidth allocated for a single voice-frequency transmission channel is usually 4 kHz, including guard bands, allowing a sampling rate of 8 kHz to be used as the basis of the pulse code modulation system used for the digital PSTN.
An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP Network such as that of a company. The phones use control protocols such as Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA).It may have many features an analog phone doesn't support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers.
Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. In telephony, the local loop is the physical link or circuit that connects from the demarcation point of the customer premises to the edge of the carrier or telecommunications service provider''s network. At the edge of the carrier access network in a traditional PSTN scenario, the local loop terminates in a circuit switch housed in an ILEC CO. Traditionally, the local loop was wireline in nature from customer to central office, specifically in the form of an electrical circuit provisioned as a single twisted pair in support of voice communications. Where the number of local loops was restricted, different customers could share the same loop, known as a party line. Modern implementations may include a digital loop carrier system segment or fiber optic transmission system known as fiber-in-the-loop. The local loop may terminate at a circuit switch owned by a CLEC and housed in a POP, which typically is either an ILEC CO or a "carrier hotel". A local loop may be provisioned to support data communications applications, or combined voice and data such as digital subscriber line.
Before now, all telephony services are based on the circuit-switched networks, known as the Public Switched Telephone Networks (PSTN). But nowadays, a new trend has emerged that provides telephony services over internet protocol networks, referred to as Voice over IP (VoIP) or IP telephony. There are a few driving force and motivational factors that are behind the VoIP network - the cost savings, the voice and data applications' integration etc are the crux of the innovation. The cost saving is achieved mainly by reduced cost of making a long distance call, maintenance, deployment, management etc., while the integrated applications of voice and data are teleconferencing, integrated voice mail and email, the call distribution intelligence and automation. Therefor, with this era of innovative business practices in information technology that involves voice and data networks integration, both transmitted over the internet and implemented on a local area network (LAN).This book is undoubtedly beneficial to Small and Medium sized businesses for convergence of data and voice networks in their existing data network to accommodate the VoIP solution in the same LAN.
Voice over IP telephony is gaining popularity quickly, and is often replacing traditional PSTN connectivity due to its low cost and easy implementation. In moving from a legacy PSTN system to a newer technology such as VoIP, security is often overlooked by vendors and users. This can present a significant problem in the corporate environment, where security in terms of both privacy and service is a key requirement. For residential users, credential theft can result in a significant annoyance, such as toll fraud or identity theft. This book focuses on attacks that compromise privacy, and details methods of detecting and mitigating these attacks. IPSec is used to ensure that signalling and media traffic are unable to be intercepted. These techniques should help administrators of networks carrying voice traffic to implement necessary measures to ensure that security and privacy remain intact. They should be particularly useful in corporate environments, where sensitive information needs to remain inaccessible to anyone other than its intended recipients.
The appearance of VoIP technology has now made feasible the use of data network for data, as well as voice communication. In this book, the protocol conversion between two different networks, packet scheduling and routing schemes for gateway selection has been introduced and also verified with Asterisk server. The recent trend for convergence of different access network technologies, market deregulation and emergence of dual-mode end-user devices allow the same destination to be reached by alternative paths and interfaces.Providing a good QoS in heterogeneous networks for irregular traffic flows remains a significant challenge. One of the difficult tasks is that of gateway location, also known as gateway selection, path selection, gateway discovery, and gateway routing. The main objective of this book is to design and develop a PSTN-IP Telephony Gateway (PITG) model and to assist routing packets while ensuring minimal call blocking probability with respect to call arrival rate or load, to increase the overall performance of the system. Specifically, the primary focus is on the development, analysis and optimization of the PITG model and evaluation of a gateway selection algorithm.
Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Yate is a telephony engine, a softswitch with PBX capabilities, originally created in 2004 by Paul Chitescu of NullTeam. It is released under a GNU General Public License with an exception for linking with OpenH323 and PWlib (licensed under MPL). Started as a free/open source softswitch for connecting PSTN with VoIP, support for other telephony technologies like conference, PBX, IVR has been added later on. During the years Yate has evolved into a Unified Communications server providing advanced integration with instant messaging, video and fax. Historically the basic function of Yate is the softswitch, connecting together call legs either to external devices or internal server resources. The logic for routing calls uses messages and can be implemented in configuration files, database queries or external programs. Call signalling is possible using PSTN protocols like ISDN or SS7, either directly attached over a TDM interface card (T1, E1, BRI), analog interface or by a remote Signaling gateway. A variety of VoIP protocols can be used without the need of special hardware: SIP, H.323, IAX2 or Jingle
Traditional telephony last about 100 years in use as the basic voice communication because of its reliability that satisfy the normal needs. Packet switching network, especially the Internet, which rapidly spread , attract more and more applications, because of its flexibility and efficiency. One of the most important applications is VoIP, which transmit voice in the same method of transmitting data, taking into account that voice is a real time application. Once we can transmit the voice over IP network, we do not need to route calls using the expensive and large central switches of PSTN, since we can route the calls in the same manner that we route data in IP network. This is the task of the soft switch, soft switch may seem to be the counterpart of central office or PBX of PSTN network. Finally, this project outlines the overall system of VoIP communication, and then show how to implement a whole telephone system based on IP protocols. The implementation of this project will improve the telephony service and management in (IUG).
Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Plain old telephone service (POTS) is the voice-grade telephone service that remains the basic form of residential and small business service connection to the telephone network in many parts of the world.The name is a retronym, and is a reflection of the telephone service still available after the advent of more advanced forms of telephony such as ISDN, mobile phones and VoIP. POTS has been available almost since the introduction of the public telephone system in the late 19th century, in a form mostly unchanged to the normal user despite the introduction of Touch-Tone dialing, electronic telephone exchanges and fiber-optic communication into the public switched telephone network (PSTN).