VoIP Learning Center
Faxing Over Satellite (FoIP)
Fax studies suggest that 1/3 of all IP faxes fail. Over satellite, fax is even more unpredictable based on the quality of the satellite link. The NETFAX satellite faxing solution uses proprietary technology to send and receive fax over any network. Successfully. 100% of the time. Despite the network being optimized for VoIP or not. Regardless of packet loss, jitter or latency.
Our solution is simple and 100% reliable. Use a standard fax machine attached to our device and press send. It really is just that simple. Never resend a fax with the Mobilsat satellite faxing solution.
We provide you with a NETFAX SFTA device that you can plug your existing fax machine into. This allows your users to use the technology they are comfortable with. Then you plug the SFTA into the nearest standard computer network port (which it can still share with the adjacent computer). Then they send and receive faxes just like they always have and you get to run the system over the satellite network.
The NETFAX SFTA provides a store and forward fax service that captures the fax image at the remote satellite location before it is sent over the satellite link. When the remote unit is transmitting a fax, it emulates the dial tone of the land line and communicates with the local fax machine using standard T.30 fax communication protocols. As the communication is all done at the remote unit, the latency and jitter associated with the Internet connection are no longer a factor, ensuring the communication to the attached fax machine is successful every time.
911 for VoIP
1. MOBIL SATELLITE TECHNOLOGIES VoIP E-911 SERVICES MAY NOT OPERATE DURING A POWER OUTAGE. In the event of a commercial power outage, the Internet Access device (IAD) will lose power causing a loss of voice and data service, including access to E-911 services. Once power service is restored, you may be required to reset or reconfigure your equipment before you will be able to use Mobil Satellite Technologies VoIP service to contact E-911 services. You are responsible for providing an uninterruptible backup power supply if you wish to ensure continued operation of electrical equipment in the event of a power outage.
2. E-911 SERVICES WILL NOT OPERATE IF YOUR BROADBAND CONNECTION IS DISRUPTED OR MOBIL SATELLITE TECHNOLOGIES VOIP SERVICE HAS BEEN SUSPENDED FOR ANY REASON, INCLUDING, FOR EXAMPLE, NON-PAYMENT. Once your broadband connection and Mobil Satellite Technologies VoIP service have been restored, you may be required to reset or reconfigure your equipment before you will be able to use Mobil Satellite Technologies service to contact E-911 services.
3. YOU MUST PROVIDE AND CONTINUOUSLY UPDATE MOBIL SATELLITE TECHNOLOGIES WITH YOUR CORRECT SERVICE ADDRESS OR YOUR E-911 SERVICES CALLS MAY BE ROUTED TO EMERGENCY PERSONNEL WHO WILL NOT BE ABLE TO ASSIST YOU. You are responsible to ensure your 911 addresses are correct. 911 emergency services use the Caller ID of your phone to dispatch emergency service to the correct address. It is imperative that the correct address is registered and the correct caller ID is sent which corresponds to the physical location. VoIP phones are easily moved to new locations and the caller ID can be modified by the customer.
4. MOBIL SATELLITE TECHNOLOGIES E-911 SERVICES CALLS MAY NOT COMPLETE OR MAY BE ROUTED TO EMERGENCY PERSONNEL WHO WILL NOT BE ABLE TO ASSIST YOU IF YOU DISABLE OR DAMAGE THE IAD.
5. MOBIL SATELLITE TECHNOLOGIES E-911 SERVICES CALLS MAY BE DELAYED OR DROPPED DUE TO NETWORK ARCHITECTURE. Due to network or Internet congestion or problems, calls to E-911 services made using Mobil Satellite Technologies VoIP services may be dropped, in which case you will not be connected to emergency services, or your E-911 calls may take longer to connect than E-911 calls made using traditional telephone service. YOU SHOULD MAINTAIN ALTERNATE MEANS OF CONTACTING E-911 SERVICES AND YOU MUST INFORM YOUR MOBIL SATELLITE TECHNOLOGIES VOIP USERS OF THESE ALTERNATE MEANS.
6. MOBIL SATELLITE TECHNOLOGIES VoIP E911 SERVICES MAY NOT BE GENERALLY AVAILABLE TO MOBILE USERS. Because of the continually moving nature of mobile (vehicle based) users, it is not always practicable for users of the Mobil Satellite VoIP service to notify Mobilsat of their new location each time they relocate as specified in Paragraph # 3 above. Mobile customers and users of the Mobil Satellite Technologies VoIP service understand and agree that unless the correct address or location information has not been provided by Customer, there will be no E-911 service provided, and that they will not be able to contact emergency services by dialing 9-1-1 using Mobil Satellite Technologies VoIP service. Customer acknowledges that it is their sole responsibility to inform all users of their Mobil Satellite Technologies VoIP services that they may not be able to contact emergency services by dialing 911 using Mobil Satellite Technologies VoIP service unless the correct location has been identified by Customer.
Mobil satellite Technologies maintains an online portal where customers can update their e-911 location information. Click here.
When talking about VoIP, especially over a satellite connection, there are several important elements that require consideration.
Latency: There is approximately 280 msec of one-way propagation delay due to the location of the Geo Stationary orbit and the speed of light. Regardless of the satellite product, this propagation delay must be considered and overcome. Today, numerous overseas calls originated in the United States are actually transmitted as VoIP over satellite calls, particularly if provided by the smaller long distance carriers.
Jitter: Quantifies the effects of network delays on packets arriving at the receiver. Packets transmitted at equal intervals from the transmitting gateway arrive at the receiving gateway at irregular intervals. Excessive jitter has the effect of making speech choppy and difficult to understand. Jitter is calculated based on the inter-arrival time of successive packets. For high-quality voice, the average inter-arrival time at the receiver should be nearly equal to the inter-packet gaps at the transmitter and the standard deviation should be low. Jitter buffers (packet buffers that hold incoming packets for a specified amount of time) are used to counteract the effects of network fluctuations and create a smooth packet flow at the receiving end.
Packet Loss: Packet loss or packet corruption will cause degradation of voice quality. Since all of the voice traffic is UDP/IP and would not be retransmitted (like in the case of TCP/IP) all packets would be completely lost if the packet becomes lost or corrupted. It is extremely important to have very low Bit Error Rates (BER) to ensure low or no corruption or loss.
QoS and Traffic Prioritization: Packet switched networks are subject to congestion as typical data traffic is bursty. Congested networks can wreak havoc on a VoIP call with delayed, dropped,or packets that are out of sequence. It is a necessity to have QoS and Prioritization in order to guarantee delivery of VoIP traffic through congested links.
Compression Technologies: There are many encoding schemes that have been standardized for voice. The most dominant standard in the industry is the G.729 codec. G.729 encoding requires 8Kbps of bandwidth, but because of the overhead associated with IP/UDP/RTP headers, the actual bandwidth needed is between 16 and 18Kbps (depending on the equipment vendor and configuration). With Compress RTP (cRTP) the total bandwidth requirement per call will drop to about 10Kbps.
Required bandwidth per VoIP call: To maintain the quality of a shared satellite network, operators must make sure that the use of VoIP makes the most efficient use of the available bandwidth, which means using the lowest Codec available and putting enough CIR onto the account. Bandwidth required per VoIP call will depend the encoding standard used, header compression, and payload size. For example, with G.729(b), payload of 40 bytes, and no header compression, a VoIP call would take about 16Kbps of bandwidth. With header compression this would require 10Kbps of bandwidth.
Internet Telephony Glossary of Terms
Analog audio signals
Analog audio signals are used to transmit voice data over telephone lines. This is done by varying or modulating the frequency of sound waves to accurately reflect the pitch of the sound. The same technology is used for radio wave transmissions.
A data communications method in which bits are sent one after the other with a start and stop bit used for flow control. This as opposed to synchronous communication where blocks of data are transmitted using a synchronizing clock.
ATA or the analog telephone adaptor is the hardware device that connects the conventional telephone to the Internet through a high speed bandwidth line, provides the interface to convert the analog voice signals into IP packets, delivers dial tone and manages the call setup.
The ITU has defined multiple audio codecs for use with H.323. All of them are also compatible with SIP, which is codec-agnostic.
G.711 is 3 kHz audio encoded at 64-kbps. G.711 is PCM audio, the format used for voice delivery over traditional telephone networks and exchanges.
G.722 is high-quality 7kHz audio in 48-, 56-, or 64-kbps streams. Two lower-quality, narrow-band revisions exist: G.722.1 encodes the audio at 24- or 32-kbps, and G.722.2 encodes at around 16kbps.
G.723.1 is used for compressing speech at very low bit rates: 5.3- and 6.3-kbps.
G.728 is 3.4kHz audio encoded at 16-kbps, but uses much smaller packet sizes (.625 millisecond, as compared to 37.5ms for G.723.1) to guarantee low delays.
G.729 is a newer voice codec using 8-kbps streams and 15ms packet sizes. There are two variations, G.729 and G.729A, that differ only in their mathematical implementation.
Speex is an open source speech codec. In contrast to the G-series codecs listed above, it is not protected by patents. It encodes at variable bitrates, from 2.15- to 44.2-kbps.
GSM6.10 is another open source codec, encoding at 13.3-kbps. At this time there is an unresolved patent dispute surrounding the codec, but is still supported by multiple software programs.
A verbal choice provided by a recording over the phone. Audio choice menus are common in automated attendant, IVR and fax-on-demand systems. They are prompts for caller input. Audio menus can instruct you to speak commands or hit touch-tones as commands.
Audio Response Unit (ARU)
A computer telephony system incorporating voice store and forward technology. There are both passive and interactive ARUs. Passive ARUs simply play out messages. Interactive ones play messages based on input from callers.
Or Audio Conferencing. The original technology used for audio teleconferencing was based on PBX conferencing circuits. Setting up conference calls through the PBX is cumbersome, voice quality degrades as the number of people on a call increases and there are capacity limitations. As a result, specialized conference bridges were developed to improve capacity and voice quality. Conference bridges, however, require trained operator intervention to schedule and invoke most features. As a result, individual corporations find the cost of ownership prohibitive, and the market for such products has been concentrated on service bureau providers. Today, PC-based systems combine the freedom of conference bridges. By installing a conference server on your voice networks, you can set up, attend, and manage your own conferences over any touch-tone telephone. Additionally, users can schedule meetings using desktop software from their e-mail systems, or from a Web browser. The latest word in this area is having the endpoints themselves being able to provide local mixing, hence eliminating the need for network based conference servers!
Bandwidth is the volume of data that can be transmitted over a communication line in a fixed amount of time. It is expressed in bits per second (bps) or bytes per second for digital devices and in cycles per second, or Hertz (Hz) for analog devices. Bandwidth can also be defined as the difference between a band of frequencies or wavelengths.
The cable modem is a device that is used to connect a computer to the high speed coaxial cable run by cable TV companies to provide access to the Internet. The connection is made through an Ethernet port, which is a shared medium and can affect download speeds if too many users log on simultaneously to the Internet on that particular cable segment. However, despite this cable modems provide extremely fast access to the net.
The time interval between when the phone is taken off the hook for a test call and when it is put back on the hook.
Circuit switched networks
These networks have been used for making phone calls since 1878. They use a dedicated point-to-point connection for each call. This reduces their utility because no network traffic can move across the switches that are being used to transmit a call.
Client (Softphone client)
The software installed in the user’s computer to make calls over the Internet.
A calling feature for inbound calls that will “roll past” a busy signal or try multiple numbers until the call is answered.
Call setup time
The length of time, measured in seconds, required to establish a circuit-switched call between users.
Class 5 (Telephony) switch
A Class 5 switch, in United States telephony jargon refers to a telephone switch or exchange located at the local telephone company’s central office, directly serving subscribers. Class 5 switch services include basic dial-tone, calling features, and additional digital and data services to subscribers using the local loop. A key part of SIP/VoIP/IMS networks/systems are IP based class 5 switches (In the IMS environment they are known as class 5 App Servers).
The loss of speech-signal components, resulting in the dropping of the initial or end parts of a word or words.
The noise on a channel or circuit with a termination but no signal (holding tone) at the transmitting end, measured through a C-message filter.
Codec is a term that arises from the Compressor-Decompressor or enCOder/DECoder process. It is used for software or hardware devices that can convert or transform a data stream. For instance, at the transmitting end codecs can encode a data stream or data signal for easy transmission, storage or encryption. At the receiving end, they can decode the signal in the appropriate form for viewing. They are most suitable for videoconferencing and streaming media solutions.
This is a term that is used to indicate the squeezing of data in a format that takes less space to store or less bandwidth to transmit. It is very useful in handling large graphics, audio and video files.
A device used to connect multiple parties over the phone. A proctor or operator can man conference bridges, or they can be supervised. There are both stand-alone conference bridges and conference bridge functions built in to some PBXs (Private Branch Exchange). These systems have circuitry for summing and balancing the energy (noise) on each channel so everyone can hear each other. More sophisticated conference bridges have the ability to “idle” the transmit side of channels of non- speaking parties. Some conference bridges use “clVoxising” to idle or reject the input of touch tones or other signals. There are VoIP based Conference Bridge servers. They may be controlled via protocols such as SIP or Megaco. they send/receive media by using the RTP protocol.
This is the process that is used to compress large data files into mall files so that they use less bandwidth during transmission and less disk space when stored. The compression depends upon the repeatable patterns of binary 0s and 1s. The higher the number of repeatable patters, the higher is the compression. The right compression codes can compress data files to 40% of their original size. The graphics files can be compressed even more – from 20% to 90%.
The time interval, measured in milliseconds, between when a phone is taken off the hook and when a dial tone sounds.
Digital Subscriber Line
A high-sped digital switched service using existing copper pairs to connect subscriber CPE (Customer Premises Equipment) to the Central Office. DSL handles more data downstream (data flowing towards the subscriber) than upstream (towards the network).
A computer program running on a web server, translating domain names into IP addresses. In the last years special types of domain names records were added to the DNS world-wide system, which provide support to SIP/VoIP (SRV/NAPTR, ENUM).
A DSL modem is a device that is used to connect one or more computers to the high speed DSL line provided by a DSL operator to gain access to the Internet. The customers use these modems to log on the net to download or transmit data. Since the DSL lines have high bandwidth capacity the data transfer speeds are very high.
Dual-tone multifrequency (DTMF)
The system used by touch-tone telephones. DTMF assigns a specific frequency (made up of two separate tones) to each key so that it can easily be identified by a microprocessor. This is basically the technology behind touch tone dialing.
The designation for the 2.048Mbps. ITU standard for Europe’s 30-channel digital telephone service. It is the European version of T-1 (DS-1). The bandwidth is divided into two signaling channels (channels 15 and 31 starting from 0) and thirty bearer (voice channels). A&B bit signaling (robbed bit signaling) is not used here. E-1 uses one of the control channels for signaling and the other for clock synchronization.
E911 is the short form of the term Enhanced 911, and is used for providing emergency service on cellular and Internet voice calls.
Echo-path delay (EPD)
The time lapse between a transmitted signal and its reflection.
Echo-path loss (EPL)
The difference in signal strength between a transmitted signal and its reflection (expressed in dB). EPL is dependent on EPD.
Emergency 911 calls
This is an emergency telephone number that handles all calls related to police, fire or medical emergencies. The number, which is allotted under the North American Numbering Plan (NANP), is answered by either a telephone operator or an emergency service dispatcher, who, in turn, alerts the appropriate emergency service.
ENUM (E.164 Number Mapping)
ENUM is a way to use the Domain Name System (DNS) for storage of E.164 numbers. More specifically, how DNS can be used for identifying available services connected to one E.164 number.
A computer based fax machine. Fax servers are “shared use” devices, typically installed on a LAN. Clients on the LAN can use the fax server from their PCs in much the same way they share a network-based (shared) printer. Faxes can be generated by users at their workstations and “printed” to the fax server for transmission. Likewise, fax servers can route incoming faxes to printers, file server directories or to individual users. Fax servers save users from having to print documents, carry them to the fax machine and subsequently wait for them to be transmitted after creating a cover page.
A feature that allows calls to find you wherever you are, ringing multiple phones (such as your cell phone, home phone, and work phone) all at once.
The duration and number of prolonged clipping events during a call, where the degraded surface of the signal falls close to zero. The ratio of frame mutes to total clipping events is displayed by the Frame Muting Ratio (%) indicator.
In data communications, a packet switching method that uses available bandwidth only when it is needed. This fast packet switching method is efficient enough to transmit voice communications with the proper network management.
In telephony and data communications, the ability for both ends of a communication to simultaneously send and receive information without degrading the quality or intelligibility of the content.
Gateway In VoIP systems
A network device that converts voice and fax calls in real time from the public switched telephone network (PSTN) to an IP network.
An ITU standard that lays down guidelines for real time voice and videoconferencing utilities on the Internet. The H.323 standard supports voice, video, data, application sharing and whiteboarding and defines media gateways for conversion to packets.
Refers to devices or deployment strategies designed to provide access to fully functioning systems at all times. One such strategy is to cluster devices so that the primary device can fail over to the secondary one if necessary.
IM, which stands for Instant Messenging, is a software that allows users to exchange messages in real time. However, to do so both the users must be logged on to the instant messaging service at the same time. Some of the popular IM services are: MSN Messenger, AOL Instant Messenger, Yahoo! Messenger, Google Talk and ICQ.
IMS stands for IP Multimedia Subsystem. It is a general-purpose, open industry standard for voice and multimedia communications over packet-based IP networks (originally defined by the 3GPP standard organization). It is a core network technology, that can serve as a low-level foundation for technologies like Voice over IP (VoIP), Push-To-Talk (PTT), Push-To-View, Video Calling, and Video Sharing. IMS is based primarily on SIP (session initiation protocol).
Interactive Voice Response IVR.
In computer telephony, Interactive Voice Response is a horizontal application wherein computer-based information is accessed over the phone – with a telephone versus a computer. An IVR platform uses computer telephony components to translate callers’ touch-tones or voice commands into computer queries after the callers hear an audio menu. For example: “Please enter your account number using the touch-tones on your telephone.” These queries are then “fetched” by the IVR platform from the host computer. In some cases, the information resides in the same platform (self-hosted). The information is then converted into voice commands and then spoken over the phone to the caller. These spoken prompts can be pre-recorded, digitized speech messages that are then concatenated to form whole sentences. For example: “Your bank balance is five hundred and sixty-three dollars”. The responses to the caller an also take the form of text-to-speech prompts. IVR systems can also be used for callers to change the information in a database instead of just “listen” to the information.
The current-day public and global computer network or “information super-highway.” The Internet is an outgrowth and combination of a variety of university and government sponsored computer networks. Federal and private sector subsidies supported the DARPA-NET. NSFnet (National Sciences Foundation) and thousands of other subnetworks, which were used to do inter-agency research and communication. Today, the Internet is made up of millions upon millions of computers and subnetworks – almost entirely supported by commercial funds except in countries where deregulation has not occurred. The internet is the substrate and chief communications backbone for the World Wide Web (WWW), the “graphical interface” of the Internet.
Internet congestion occurs when a large volume of data is being routed on low bandwidth lines or across networks that have high latency and cannot handle large volumes. The result is slowing down of packet movement, packet loss and drop in service quality.
Internet Telephony (AKA IP Telephony)
Any means of transmitting the human voice (real time or close to real time) over the internet. There are several components: 1) On the client side, a multimedia-equipped PC with special client software will digitize your voice. This can be done with a voice modem or other voice encoding method; 2) A direct or dial-up connection to the internet allows your voice to be transmitted in packet form to its destination; 3) Connection with the far side is achieved by IP address search, common servers or beacons to identify the called party (and to “ring” that person’s phone); 4) A similar arrangement on the far end completes the call and allows both parties to speak. There are also PSTN/Internet gateways that allow regular telephone callers to make Phone-to-Internet-to-Phone connections. There are PC-to-Phone connections and Phone-to-PC connections.
IP, which is the acronym for Internet Protocol, defines the way data packets, also called datagrams, should be moved between the destination and the source. More technically, it can be defined as the network layer protocol in the TCP/IP communications protocol suite.
An IP address, also known as Internet Protocol address, is the machine number used to identify all devices that are connected to the net. Each device has its own unique number which it uses to communicate. This number is fixed in the case of those computing devices that have a fixed IP address. The rest are allotted a dynamic IP address, which is valid for the period they are connected to the net. The numbers range from 0.0.0.0 to 255.255.255.255.
IP mapping is the process of identifying IP addresses on the basis of their geographical locations. The mapping enables web administrators to pinpoint the location of any computing device connected to the Internet.
IP Phone (AKA Internet Phone or SIP Phone or VoIP Phone (or H.323 Phone))
An IP phone is one that converts voice into digital packets and vice versa to make phone calls over Internet possible. It has built-in IP signaling protocols such as SIP or H.323 that ensure that the voice is routed to the right destination over the net. On the media side the IP Phone uses audio or/and video codecs such as G.711 or/and H.261 respectively over RTP. The IP phones come with several value added services like voicemail, e-mail, call number blocking etc.
Internet Service Provider. A business that provides subscriber-based access to the Internet. Subscribers can be individuals or businesses. According to Jack Rickard, publisher of Boardwatch Magazine, ISPs operate at the fourth or lowest level of the Internet. At the third level, regional providers aggregate traffic from lower-order ISPs to the second, backbone level. The highest level in North America is the NAP (Network Access Point), which act as peer-to-peer interconnection points for the largest backbones. There are three “official” NAPs located in San Francisco, Chicago and Pennsauken, New Jersey. ISPs use both Internet Routers, Servers and Rack-Mounted modems to provide a variety of services including Web Site hosting, FTP service, e-mail accounts, unified messaging, audio and video broadcasting and in some cases – Internet Telephony and Fax Gateway service.
ITU, which is the acronym of International Telecommunication Union, is a telecommunications standards body based in Geneva. It works under the aegis of the United Nations and makes recommendations on standards in telecommunications, information technology, consumer electronics, broadcasting and multimedia communications.
It is a term used to indicate a momentary fluctuation in the transmission signal. This happens in computing when a data packet arrives either ahead or behind a standard clock cycle. In telecommunication, it may result from an abrupt variation in signal characteristics, such as the interval between successive pulses.
Kbps is the acronym for kilobits per second and is used to indicate the data transfer speed. If the modem speed, for instance, is 1 Kbps then it means that the modem can route data at the speed of one thousand bits per second.
Lag is the term used to indicate the extra time taken by a packet of data to travel from the source computer to the destination computer and back again. The lag may be caused by poor networking or by inefficient or excessive processing.
Latency is the time that elapses between the initiation of a request for data and the start of the actual data transfer. This delay may be in nanoseconds but it is still used to judge the efficiency of networks.
Mean opinion score (MOS)
A measurement of the subjective quality of human speech, represented as a rating index. MOS is derived by taking the average of numerical scores given by juries to rate quality and using it as a quantitative indicator of system performance.
MEGACO/H.248 (RFC 3525 defines version 1 (replaced MGCP))
This is the latest industry standard protocol for interfacing between hosts and call agents called Media Gateway Controllers (MGC’s) and Media Gateways (MG’s) – eg. an IP Telephone and the PSTN. The standard is the result of a unique collaborative effort between the IETF and ITU standards organizations. Derived from MGCP (which, in turn, was derived from the combination of SGCP and IPDC).
In computer telephony, any means of message store and forward. This includes fax mail, voice mail and broadcast messaging. This horizontal application is the most popular of all other voice solutions. Messaging systems provide for the store and forward of “non-real time” communication. For example, a recorded voice message can be stored for later play back either locally or remotely, or a fax can be received and stored before it is re-transmitted to the ultimate recipient. Messages, then, can vary in content and media type – the distinction being that they are recorded or stored for pick up in the future. The time between original storage and retrieval of a message can be created and stored by a sales manager for later retrieval by multiple (worldwide) sales people. The sales staff can listen to the message at different times over an extended period. This is due to the nature if random retrieval by the recipients in their respective time zones. Messaging systems are a kind of “shared tenant” answering machine, because messages that were intended for as many as a thousand or more users can be stored and controlled by the same system. If a community of users agree on some basic ground rules, messages can be shared, forwarded, and distributed to multiple recipients in the same fashion as e-mail.
MGCP (later was replaced by Megaco/H.248)
Media Gateway Control Protocol; RFC 2705 – is worth mentioning. It is an in-development IETF standard for converting voice signals from the conventional telephone network into data packets (and vice-versa), and may be used in conjunction with SIP or H.323. As its name suggests, it is used mainly by Media Gateway Controllers to control Media Gateways.
Short for Modulator/Demodulator. Equipment that converts digital signals to analog signals and vice-versa. Modems are used to send data signals (digital) over the telephone network, which is usually analog. A modem modulates binary signals into tones that can be carried over the telephone network. At the other end, the demodulator part of the modem converts the tones to binary code.
Stands for North American Numbering Plan. A telephone numbering system that has evolved the way area codes and numbers are allotted. The system was established in 1947 and covers the United States, Canada and a few neighboring areas. It uses a three-digit area code and seven-digit telephone numbers. Its fiat is, however, limited to the public switched telephone networks only.
A logically grouped unit of data. Packets contain a payload (the information to be transmitted), originator, destination and synchronizing information. The idea with packets is to transmit them over a network so each individual packet can be sent along the most optimal route to its. Packets are assembled on one end of the communication and re-assembled on the receiving end based on the header addressing information at the front of each packet. Routers in the network will store and forward packets based on network delays, errors and re-transmittal requests from the receiving end.
Packet loss is the term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latency or on account of overloading of switches or routers that are unable to process or route all the incoming data.
A means of economically sending and receiving data over alternate, multiple network channels. The premise for packet switching is the packet, a small bundle of information containing the payload and routing information. Packet switching takes data, breaks it down into packets, transmits the packets and does the reverse on the other end. Packets can be sent in order and then be received in a different order – only to be put back in the correct order in seconds. There are slow packet switching networks, like the old SNA networks – and there are fast packet networks based on Frame Relay and ATM. Although traditionally used for data, packet networks, especially well-managed ones, are becoming suitable for real-time transmission of voice and video.
Private Branch Exchange. Or PABX (Private Automatic Branch Exchange). In telephony, a PBX system behaves as a customer’s premises over trunk lines (thus the term “branch”). At first, PBXs mimicked a small telephone company switchboard. Users would use an operator to take and make telephone calls to and from the PSTN (Public Switched Telephone Network). Over time, users were able to dial directly, without the use of an operator. Today, computer telephony platforms such as automated attendants are able to route incoming calls automatically, too.
The term peer-to-peer is used to indicate a form of computing where two or more than two users can share files or CPU power. They can even transmit real time data such as telephony traffic on their highly ad hoc networks. Interestingly, the peer-to-peer network does not work on the traditional client-server model but on equal peer nodes that work both as “clients” and “servers” to other nodes on the network.
Point of Presence, equivalent of a local phone company’s central office. The place your long distance carrier terminates your long distance lines just before those lines are connected to your local phone company’s lines, or to your own direct hookup.
Alternate Definition: Post Office Protocol. An Internet standard for the storage and retrieval of email messages.
The time interval between when the caller presses the last digit of a number and when the phone on the other end begins to ring. It is the basic quantifier for routing speed as perceived by the user.
POTS (plain old telephone service)
The typical, familiar model of a single phone line and a single phone number.
It is a convention or standard that defines the procedures to be adopted regarding the transmission of data between two computing end points. These procedures include the way the sending device should sign off a message or how the receiving device should indicate the receipt of a message. Similarly, the protocols also lay down guidelines for error checking, data compression, and other relevant operational details.
Public Switched Telephone Network. The combination of local, long-distance, and international carriers that make up the worldwide telephone network.
QoS (quality of service)
The ability of a network (including applications, hosts, and infrastructure devices) to deliver traffic with minimum delay and maximum availability.
A communication wherein any perceptible delay between the sender and receiver are minimal and tolerated. Regular telephone calls are real time. Point-to-point fax transmissions are “close” to real time. Voice messaging is in non-real time.
The designation for connecting a tip and ring circuit to a standard, modular, six-position jack. The green and red wires go in the middle (only) pair, and the outside positions of the connector are unused.
Eight-position modular connector used for data transmission over standard twisted or flat pairs.
A router is a network device that that handles message transfer between computers that form part of the Internet. The messages, which are in the form of data packets, are forwarded to their respective IP destinations by the router. A router can also be called the junction box that routes data packets between computer networks.
This is a methodology used to measure the value of an analog signal at regular intervals, and encoding it into a digital format for VoIP phone services.
An addressable entity providing application and administrative support to the client environment by responding to client requests and maintaining the operational integrity of the server.
Signaling System #7
Or SS7. The basis for modern methods to route traffic with out-of-brand signaling. Its forerunner, CCIS (Common Channel Interoffice Signaling), used 4.8 Kbps data links to transmit call set up and tear down messages to switching office adjunct computers and packet switches. SS7 in itself is not a network service offering, but rather the underlying infrastructure with which many existing and proposed offerings are based. For example, local Basic Rate ISDN (BRI) services can tap into SS7, so 64 Kbps packetized data can be routed with the help of the network’s out-of-band signaling capability. In addition, nationwide Primary Rate ISDN (PRI) services can use the same backbone.
SIP (Session Initiation Protocol)
An Internet Engineering Task Force (IETF) standard for initiating, maintaining, and terminating an interactive user session involving video, voice, chat, gaming, virtual reality, and more.
SIP phone (Also see above IP Phone)
A SIP phone is a telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet (for signaling (and uses RTP for media)). The SIP phones come with several value added services like voicemail, e-mail, call number blocking etc. There are (normally) no charges for making calls from one SIP phone to another, and negligible charges for routing the call from a SIP phone to a PSTN phone.
Skype is a peer-to-peer Internet telephony company that revolutionized the way voice calls are made by using VoIP technology. The company, which has been acquired by eBay, was founded by Niklas Zennström and Janus Friis. Skype users can speak to other Skype users for free, but have to pay a small fee for calling or receiving calls from conventional phones.
IP telephony software that lets users send and receive calls from non-dedicated hardware, such as a PC or Pocket PC device. It is typically used with a headset and microphone.
Note: Soft Phone and SIP Phone might be (but not necessarily) special cases of each other.
It is a software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks. This software is loaded in computers and is now replacing hardware switches on most telecom networks.
The measure of the strength of a received voice signal.
Speech recognition describes a group of special technologies that allow callers to speak words, phrases, or utterances that are used to control applications. In the case of voice processing, speech recognition is used to replace touch-tone input, make for more intuitive menu structures, and ad a level of simplicity and security to some systems. Speech recognition, on the other hand, is a technology that uses the spoken word as input that has an effect on the logic flow and execution of the program in question.
Store And Forward
As the name implies, the discipline of storing a message or transmission for later playback or transmission. As opposed to real time communication, store and forward is the basis for all messaging systems including email, fax-on-demand, unified messaging, etc. In data communications, store and forward applies to momentary buffering of packets or other data strings.
North American digital standard for high-capacity transmission of telephony and data communications. In telephone T-1 provides a 1.544 Mbps link which is broken down in to 24 discrete, 64 Kpbs voice-grade channels. In data communications, T-1 links are used to directly connect CPE (Customer Premises Equipment) routers to the Internet and for Private Data Network or VPN circuits.
North American standard for DS-3. Operates at a signaling rate of 44.736 Mbps, or the equivalent of 28 T-1s.
Transmission Control Protocol. The transport layer protocol developed for the ARPAnet which comprises layers 4 and 5 of the OSI model. TCP controls sequential data exchange in TCP/IP for remotely hosts in a peer-to-peer network.
Taken from Greek root words meaning “far sound”, telephony is the discipline of converting or transmitting voice or other signals over a distance, and then re-converting them to an audible sound at the far end.
A multi-user, multi-tasking operating system originally developed in 1969 by Ken Thompson of AT&T Bell Laboratories. UNIX is used in telephone company and mission critical applications.
There are fewer video codecs (than audio codecs) associated with the H.323 and SIP protocol suites (thankfully).
H.261 is a video codec use for wideband (>= 64Kbps). H.263 is used for narrowband (< 64-kbps). Both are widely supported.
H.264 is a newer narrowband codec that produces higher-quality results than H.263 and is recommended in its place. H.264 is also known as ISO 14496-10 and as MPEG-4 part 10 and as MPEG-4 AVC (Advanced Video Coding).
An application of store and forward wherein telephone access to private messages are retrieved by users for playback. Imagine a shared tenant answering machine that handles multiple telephone lines and can record incoming messages for hundreds of people simultaneously. Imagine the intended parties being able to retrieve these messages over the phone with simple touch-tone commands. Imagine full security, so no one can pick up anyone else’s messages without a special, private access code. That’s voice messaging. Voice messaging systems take many forms. There are CPE (Customer Premises Equipment) versions and Service Bureau or Telco versions. The basic idea is the non real-time sending and receiving of private messages. Some systems support the broadcast of messages to multiple recipients. Some provide message waiting notification via pager, message waiting light or “outdial” telephone calls.
VoIP (Voice over IP)
The process of making and receiving voice transmissions over any IP network. IP networks include the Internet, office LANs, and private data networks between corporate offices. The main advantage of VoIP is that users can connect from anywhere and make phone calls without incurring typical analog telephone charges, such as for long-distance calls.
VoIP closed systems (as opposed to Open Standards (such as SIP, H.323 or MGCP)
AOL, Yahoo, and Apple all offer “voice chat” capability via their instant messaging networks. These systems are closed and for the most part unable to interoperate with other, standards-based products and may use undocumented protocols. However, even where the protocols are known or have been reverse-engineered, the audio codecs are proprietary. It is known that AOL’s voice chat uses codecs from Qualcomm, Yahoo’s uses TrueSpeech from DSP Group, and Apple’s uses PureVoice QCELP.
The popular Skype service is similarly closed. It is known to use three audio codecs: iLBC (Internet Low-Bitrate Codec) and iSAC (Internet Speech and Audio Coder) from GlobalIPSound, and a third as-yet unidentified codec. The Skype protocols have not been reverse-engineered.
This device provides the conversion interface between the public switched telephone network (PSTN) and an IP network for voice and fax calls. Its primary functions include: voice and fax compression/decompression, packetization, call routing and control signaling. It also provides an interface to Gatekeepers or Softswitches, billing systems, and network management systems.
VoIP PBX, which stands for Voice over Internet Protocol Private Branch eXchange, is a telephone switch that converts IP phone calls into traditional circuit-switched TDM connections. It also supports traditional analog and digital telephones.
A VoIP phone is one that uses the Internet to route voice calls by converting the voice data into IP packets and vice versa. The phones come with built-in IP signaling protocols such as H.323 or SIP that help in the routing of data to the right destination. A VoIP phone can also be a software application that is installed in the user’s PC. In this case it is known as the Softphone. Also, the calls in this case have to be made from the PC, and not through a telephone instrument.
Client software used to view information on Web servers. Can display graphics. Web browsers are also packaged with email clients, newsreaders and in some cases, IP Telephony clients.
Web-Enabled Call Center
Any call center whose “callers” can establish a traditional of Internet-Based phone call with an agent initiated via Web Browsing Interaction. Imagine this: You cruise to a Web Page and see a product you’d like to buy. You click on a button that says “speak to a live agent”. A form pops-up and you’re prompted to enter your phone number. A few moments later your phone rings. It’s an agent from the call center associated with the Web Page you just visited.
On the World Wide Web, a server dedicated to storing data (such as Web pages in HTML format) and distributing it to Web Browsing users. Web browsers are able to download video, text, still images and audio from Web Pages. Some servers support Unified Messaging.
The noise level measured on a wideband channel in the absence of a signal.
An area where a wireless access point enables users carrying wireless-enabled laptops to log on to the Internet. The limiting condition is that the access point is configured to broadcast its presence and does not require authorization for access. Generally, WiFI hotspots are located in public places like airports, train stations, libraries, marinas, convention centers, coffee shops and hotels.
A WiFI phone is one that enables users to make phone calls from public WiFi hotspots or residential WiFI network environments. Besides voice calls, these phones can be used to send e-mails wirelessly.