Glossary
From HostedSwitch®
A B C D E F G H I J K L M N O P Q R S T U V W X Y Z
Abandoned Call
Abandoned call is a call that is concluded after a call has been answered, but before discourse has begun.
Abbreviated Dialing
Abbreviated dialing allows users to place an outbound call within their existing phone network using two or three digit codes instead of the typical seven or ten-digit codes dictated by the North American Number Plan.
ACD
ACD - Average call duration is in seconds. ACD equals to Sum of Call Durations divided by Number of completed Calls for a specified time period.
ACELP
Algebraic Code Excited Linear Prediction.
Adapter
A device that enables something to be used in a different way from which is was intended, or makes different pieces of equipment compatible. ADPCM Adaptive Differential Pulse Code Modulation.
AGPL
The GNU Affero General Public License is a free, copyleft license for software and other kinds of works, specifically designed to ensure cooperation with the community in the case of network server software.
AHCIET
AHCIET is the Ibero-American Association of Research Centers and Telecommunication Enterprises.
Founded in 1982, AHCIET is a non-profit organization. More than 50 Latin American and Spanish telecommunications companies have joined us with the sole purpose of contributing to telecommunications development as a basis of very nation’s growth.
AHT (Average Hold Time)
The average length of time between the moment a caller finishes dialing and the moment the call is answered or terminated.
AMR-WB
Adaptive Multi Rate – WideBand.
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.
ANI
Automatic Number Identification: a telephone function which transmits the billing number of the incoming call (Caller ID, for example).
Apache License
The Apache License is a free software license authored by the Apache Software Foundation (ASF). The Apache License requires preservation of the copyright notice and disclaimer.
API
API is a specified set of rules and requirements that software programs can utilize in order to establish communication with each other. Strictly speaking, APIs function to interface between various software programs in order to facilitate interaction.
Hostedswitch® API is a set of predefined methods and an environment to call these methods used to facilitate usage of the Hostedswitch routing/billing system by means other than Hostedswitch® web interface, such as member's proprietary accounting systems, web robots, etc.
AS (Autonomous System)
A set of routers under a single technical administration. An AS uses an internal gateway protocol and common metrics to route Packets within the AS, and uses an external gateway protocol to route packets to other ASs.
ASP (Application Service Provider)
An independent, third-party provider of software-based services delivered to customers across a wide area network (WAN).
ASR (Answer-Seizure Ratio)
The ratio of successfully connected calls to attempted calls (also called 'Call Completion Rate'). ASRs vary by routes. A typical ASR to Pakistan is lower than that of Germany. Reasons for this include the quality of the network and the fact that it's less likely that a call to Pakistan will encounter a device such as an answering machine. Built-in HostedSwitch®QoS Management Tools track the ASRs for all termination facilities that receive calls routed through the Hostedswitch® Softswitch.
Asterisk
An Open Source Linux-based PBX project available under the Gnu Public License. See http://www.asterisk.org
Asynchronous Communication
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.
Asynchronous Transfer Mode (ATM)
The international standard for cell relay in which multiple service types (such as voice, video, or data) are conveyed in fixed-length (53-byte) cells. Fixed-length cells allow cell processing to occur in hardware, thereby reducing transit delays.
ATA
ATA or the analog telephone adaptor is a 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.
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.
Automated Attendant
An automated attendant is an interactive voice response (IVR) application that replaces the role of a human attendant by answering incoming telephone calls with a recorded or synthesized greeting or message. The application then leads the caller through a series of menu choices to determine and route the incoming call to the proper extension, voice mailbox, service, or department.
A-law
A-law is a lossy method of audio encoding that compresses 16-bit linear PCM audio samples into 8-bit samples thus reducing bitrate by 50%.
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Backbone
A very-high-speed network spanning the world from one major metropolitan area to another. Such networks are typically provided by national Internet service providers (ISPs). Local ISPs connect to the backbone in order to transport data.
Backup Switch (Reserved Switch)
Backup Switch, also known as Reserved Switch, is a VoIP softswitch solution that is used in case of a downtime or overload of the main softswitch facilities.
Bandwidth
The maximum data carrying capacity of a transmission link. For networks, bandwidth is usually expressed in bits per second (bps).
Bargein
In an automated telephone system, the bargein feature allows users to move rapidly through prompts, interrupting the system to quickly navigate to the next prompt.
Billing Increment
A call duration measurement unit, expressed in seconds. Calls that involve a fraction of a billing increment are rounded up.
Bit
A bit is a single binary digit. Unlike a single decimal digit representing a value of 0 through 9, a bit holds a value of either 0 or 1. Similarly, while decimal numbers are, implicitly, base 10, binary numbers, which are composed of many bits, are base 2.
Bit Stuffing
In telecommunications, bit stuffing is the insertion of data bits that do not carry information. Bit stuffing is a practice utilized in data transmission and serves various purposes, including synchronizing bit streams that have dissimilar bit rates, or to fill buffers or frames.
Blink
Blink is a SIP client for Mac that can be used with any SIP provider or its own fully-featured SIP service.
Broadband
It is a term used to define high speed Internet connection, generally provided by cable TV, DSL or dedicated telecom lines. The high speeds are achieved by the carrying capacity of the cable that can carry multiple messages simultaneously.
BSD License
The BSD licenses are a family of permissive free software licenses. The original license was used for the Berkeley Software Distribution (BSD), a Unix-like operating system after which it was named.
Buyer Tariff
The price at which a VoIP provider Member can send minutes to the destination associated with the Ordered Contract. Buyer Tariff equals the sum of the Seller Tariff and the Clearing Fee. Potential Buyers can indicate Tariffs that they are willing to pay for Termination Services in their Requests.
Buyer
A Registered User or a Member of VoIP provider that intends to purchase Termination Services.
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Call
Establishment of (or an attempt to establish) voice connection between two endpoints, or between two points which provide a partial link (e.g. a trunk) between two endpoints.
Call Agent
The intelligent and controlling entity in an MGCP based IP Telephony network.
Call Completion Rate
The term, call completion rate, refers to the total number of successfully completed inbound or outbound calls versus the total number of calls that were placed or received.
Call Duration
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.
Call Flow
The setup and tear down process and steps for a call to start till finish.
Call Forwarding
A phone service feature that allows the customer to forward their phone to another phone number (for example, you can forward your home phone number to your cell number if you know you are going to be away from home).
Call Hunting
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.
Call Transfer
In telecommunications, call transferring is a technology that allows a party to transfer an ongoing call along to an attendant, extension, IVR system, or a message recording service.
Call Waiting
A phone service feature that notifies a telephone user that another incoming call is waiting to be answered. This is typically provided by a short tone on the phone, or by use of the caller ID feature. This would be a typical feature included with your VoIP Phone Service.
Caller Identification (ID)
A phone service feature that allows the recipient of a phone call to see the phone number of the originating caller (person). This would be a typical feature included with your VoIP Phone Service.
Capacity
The maximum information carrying ability of a communications facility or system.
Call Detail Record (CDR)
Information regarding a single call collected from the switch and available as an automatically generated downloadable report for a requested time period. The report contains information on the number of calls, call duration, call origination and destination, and billed amount. Hostedswitch® Members use CDR reports to bill retail customers and settle with their partners on a wholesale level.
CDP
The Cisco Discovery Protocol (CDP) is a proprietary Data Link Layer network protocol developed by Cisco Systems. It is used to share information about other directly connected Cisco equipment, such as the operating system version and IP address. CDP can also be used for On-Demand Routing, which is a method of including routing information in CDP announcements so that dynamic routing protocols do not need to be used in simple network.
Cell Relay
In telecommunications technology, cell relay describes the process used for transferring data in fixed length packets (referred to as cells).
Channel
In telecommunications networking, a communications channel acts as both a physical transmission medium (ex. the physical connection between initiating and terminating nodes of a circuit) or as a path for conveying signals and data from one point to another (ex. electrical or electromagnetic signals).
Codec
Compression-decompression. In VoIP it is a voice compression-decompression algorithm that defines the rate of speech compression, quality of decompressed speech and processing power requirements. The most popular codecs in VoIP are ITU-T G.723.1 and G.729 (AB). Examples: G.711, G.721, G.722, G.722.1, G.722.2, G.723.1, G.726, G.727, G.728, G.729, iLBC, LPC, Speex.
Competitive Local Exchange Carrier (CLEC)
A company that builds and operates communication networks in metropolitan areas and provides its customers with an alternative to the local telephone company.
Compression Delay
The delay caused by the compression of data.
Compression
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.
Congestion
The situation in which there are too many packets present in the network, leading to performance degradation.
cRTP
Compressed Real Time Transport Protocol.
CS-ACLEP
Conjugate-Structure Algebraic-Code-Excited Linear-Prediction.
C7
Common Channel Signaling 7. It is a telecommunications protocol suite defined by the ITU-T which is used by the telephone companies for interoffice signalling SS7 uses out of band or common-channel signalling (CCS) techniques. SS7/C7 uses a separated packet-switched network for the signalling purpose. SS7 is known as C7 outside North America.
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Data Communications
The transmission and reception of data between locations. Data communications require a combination of hardware (terminals, modems, multiplexers, and other hardware) and software.
Data Compression
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%.
Decompression
Process by which the full data content of a compressed file is restored.
Delay
This is always measured in seconds or fractions of seconds. There are ddelays through every electronic device even if it's only nano seconds. 1ns is 0.000000001 seconds. In a system like the internet delays are caused by the electronics(this is usually negligible), queuing delays, transmission distances and software delays.
DID
DID (DDI) in the new VoIP World. Let's say you buy a phone line from Vonage or some other phone service provider who offers phone service over broadband. The number that they provide to you, in technical terms is a DID number. This is the number that they have assigned to you to connect you to the old PSTN Networks around the world.
Dial-Peer
An addressable call endpoint -- a software structure that binds a dialed digit string to a voice port or IP address of the destination gateway. Several dial peers always exist on each router in the network, and at least two will be involved in making a call across the network, one on the originating end and one on the terminating end. In Voice over IP, there are two kinds of dial peers: POTS and VoIP. VoIP peers point to specific VoIP devices.
Dial-Peer Hunting
Process when the originating router tries to establish call on different dial peers if the originating router receives a user-busy invalid number or an unassigned-number disconnect cause code from a destination router.
Dial-Tone Delay
The time interval, measured in milliseconds, between when a phone is taken off the hook and when a dial tone sounds.
Diameter (protocol)
Authentication, Authorization and Accounting protocol for computer networks, and an alternative to RADIUS.
DNIS
DNIS stands for Dialed Number Identification Service. It is the number that the caller dialed. The DNIS, along with the ANI, are delivered to the IVR by the telco carrier for every phone call.
Domain Name System (DNS)
A domain name system is an online distributed database system that translates domain names into numerical values that are then readable by networking equipment for the purpose of locating and addressing devices worldwide.
DTMF
Dual-Tone Multifrequency; The type of audio signals generated when you press the buttons on a touch-tone telephone.
Due Date
Due Date Deadline of payment against a particular statement.
Duplex
Duplex communication is the transmission of voice and/or data signals that allows simultaneous 2-way communication.
Dynamic Jitter Buffer
Collects voice packets, stores them, and shifts them to the voice processor in evenly spaced intervals to reduce any distortion in the sound.
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E.164
The international public telecommunication numbering plan. An E.164 number uniquely identifies a public network termination point and typically consists of three fields, CC (country code), NDC (national destination code), and SN (subscriber number), up to 15 digits in total.
E1
A wide-area digital transmission scheme (European): 2,048 Mbits/s; 31 channels, 64 Kbps each.
E911
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.
Ekiga
Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet.
Elastix
Elastix is an Open Source Sofware to establish Unified Communications. Elastix's goal is to incorporate all the communication alternatives, available at an enterprise level, into a unique solution.
Empathy
Empathy is a default GNOME application for voice and video calls over SIP.
Encapsulation
In computer networking, encapsulation is a method of designing modular communication protocols in which logically separate functions in the network are abstracted from their underlying structures by inclusion or information hiding within higher level objects.
Endpoint
SIP or H.323 terminal or Gateway. An endpoint can Call and be Called. It generates and terminates the information stream.
ETSI
The European Telecommunications Standards Institute (ETSI) produces globally-applicable standards for Information and Communications Technologies (ICT), including fixed, mobile, radio, converged, broadcast and internet technologies.
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Failed Call
An attempted Call that does not receive the Connect message or SIP 200 OK message. Such calls are not billed.
Federal Communications Commission (FCC)
The Federal Communications Commission (FCC) is a U.S. government board that consists of five presidential appointees who are tasked with regulating all national telecommunications, as well as regulating international communications that either originate or terminate within the United States.
Find-me/Follow-me
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.
Firewall
A system designed to prevent unauthorized access to or from a private network. Firewalls can be implemented as hardware, software, or a combination of both. All messages entering or leaving the intranet pass through the firewall, which examines each message and blocks those that do not meet sthe security criteria specified on the firewall.
Flac
FLAC (Free Lossless Audio Codec) is a lossless audio format.
Frame Mutes
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.
FreeSWITCH
FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.
FSF
FSF (Free Software Foundation) is a nonprofit with a worldwide mission to promote computer user freedom and to defend the rights of all free software users.
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G.7xx
A family of ITU standards for audio compression.
G.711
An ITU-T PCM half- duplex codec that uses either A-law or U-law compression (64 Kbps, high quality, minimum processor load).
G.721
ITU-T specification of audio codec.
G.722
ITU-T specification of audio codec.
G.722.1
ITU-T specification of audio codec.
G.722.2
ITU-T specification of audio codec.
G.723.1
An ITU-T double rate CELP codec (6.4/5.3 Kbps, medium quality, high processor load).
G.726
An ITU-T ADPCM wave form codec (16/24/32/40 Kbps, good quality, low processor load).
G.727
ITU-T specification of audio codec.
G.728
An ITU-T low delay CELP codec (16 Kbps, medium quality, very high processor load).
G.729
An ITU-T ACELP codec (8 Kbps, medium quality, high processor load).
Gatekeeper
The central control entity that performs management functions in a Voice and Fax over IP network and for multimedia applications such as video conferencing. Gatekeepers provide intelligence for the network, including address resolution, authorization, and authentication services, the logging of Call Detail Records, and communications with network management systems. Gatekeepers control bandwidth, provide interfaces to existing legacy systems, and monitor the network for engineering purposes as well as for real-time network management and load balancing.
Gateway
In IP Telephony, a network device that converts voice and fax calls, in real time, between the public switched telephone network (PSTN) and an IP network. The primary functions of an IP gateway include voice and fax compression/ decompression, packetization, call routing, and control signaling. Additional features may include interfaces to external controllers, such as Gatekeepers or Softswitches, billing systems, and network management systems.
Generic Traffic Shaping (GTS)
A mechanism to control the traffic flow on a particular interface.
GKTMP (Cisco Gatekeeper Transaction Message Protocol)
A proprietary Cisco protocol used for communication between the Cisco IOS Gatekeeper and external applications.
GPL
The GNU General Public License (GNU GPL or simply GPL) is the most widely used free software license. The GPL is the first copyleft license for general use, which means that derived works can only be distributed under the same license terms. Under this philosophy, the GPL grants the recipients of a computer program the rights of the free software definition and uses copyleft to ensure the freedoms are preserved, even when the work is changed or added to. This is in distinction to permissive free software licenses, of which the BSD licenses are the standard examples.
Grace Period
The time interval at the beginning of a call, measured in seconds, that is not billed. HostedSwitch® Fee and Routing Fee do not apply to the grace period.
GSM (Global System for Mobile Communications)
GSM (Global System for Mobile Communications, originally Groupe Spécial Mobile), is a standard set developed by the European Telecommunications Standards Institute (ETSI) to describe technologies for second generation (2G) digital cellular networks. Developed as a replacement for first generation (1G) analog cellular networks, the GSM standard originally described a digital, circuit switched network optimized for full duplex voice telephony. The standard was expanded over time to include first circuit switched data transport, then packet data transport via GPRS (General Packet Radio services). Packet data transmission speeds were later increased via EDGE (Enhanced Data rates for GSM Evolution) referred as EGPRS. The GSM standard was further improved after the development of third generation (3G) UMTS standard developed by the 3GPP. GSM networks will evolve further as they begin to incorporate fourth generation (4G) LTE Advanced standards.
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H.225
Protocols (RAS, RTP/RTCP, Q.931 call signaling) and message formats for H.323.
H.235
For security in H.323 network.
H.239
For dual stream use in videoconferencing.
H.245
A protocol for capability negotiation, messages for opening and closing channels for media streams, etc. (i.e. media signaling).
H.246
ITU-T specification for H.323/PSTN Interworking.
H.248
ITU-T standard for a centralized VoIP network. (Same as Megaco defined by IETF.)
H.261
Used primarily in older videoconferencing and video telephony products.
H.263
Used primarily for videoconferencing, video telephony, and internet video.
H.264
Also known as MPEG-4 Part 10, or AVC (for Advanced Video Coding).
H.323
An ITU-T "umbrella" of standards for Packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoints, Gateways, Multipoint Conferencing Units (MCUs), and Gatekeepers -- and their interaction. This standard is used for many VoIP applications, and is heavily dependent on other standards, mainly H.225 and H.245.
H.350
Directory Services Architecture for Multimedia Conferencing.
H.360
An architecture for end-to-end QoS control and signaling.
H.450
For supplementary services such as call waiting, call forwarding, etc.
H.460.x
Supplements in H.323.
H.501
Protocol for mobility management and intra/inter-domain communication in multimedia systems.
H.510
Mobility for H.323 multimedia systems.
Hairpin
To send a call back in the direction that it came from.
High-Availability
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.
Hop off
Point at which a call transitions from H.323 to non-H.323, typically at a gateway. "be hopped-off locally" means "be hairpinned". Example from documentation: "If the called address does not match any known zone prefixes, the gatekeeper will attempt to hairpin the call out through a local gateway."
Hostedswitch®
HostedSwitch® is a cutting-edge class 4 VoIP switch for small and medium-size telephony providers and VoIP minutes traders. HostedSwitch® is a registered service mark of MGCP, Inc.
Hostedswitch® Billing Plan
Hostedswitch® Billing Plan is a monthly subscription plan for Hostedswitch® Members. A Billing Plan consists of Monthly Fixed Fee, Included Minutes, and Price per Additional Minute. In order to use our cloud VoIP softswitch a Member should prepay a Monthly Fixed Fee. After that a Member may send VoIP traffic through the switch up to a number of Included Minutes. If the traffic exceeds the number of Included Minutes, every exceeding minute will be billed at a Price per Additional Minute. Billing Plans are available here.
For example, if a Member selects the Plan for $250/month with 100,000 minutes included and $0.0025/per additional minute, then the Member should prepay $250 before a new billing period starts. After prepaying this amount, a Member can send up to 100,000 minutes through the switch with no additional charge. Every minute starting from 100,001th minute will be charged from Member's Balance at $0.0025 rate.
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IAX
Inter-Asterisk-eXchange protocol.
IETF (Internet Engineering Task Force)
One of two technical working bodies in the Internet Activities Board. The IETF meets three times a year to set technical standards for the Internet.
iLBC
Internet Low Bitrate Codec (iLBC) is an open source royalty-free narrowband speech codec. It is available under an open source (3-clause BSD license) license as a part of the open source WebRTC project. It is suitable for VoIP applications, streaming audio, archival and messaging.
IMS (IP Multimedia Subsystem)
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is an architectural framework for delivering Internet Protocol (IP) multimedia services. It was originally designed by the wireless standards body 3rd Generation Partnership Project (3GPP), as a part of the vision for evolving mobile networks beyond GSM. Its original formulation (3GPP Rel-5) represented an approach to delivering "Internet services" over GPRS. This vision was later updated by 3GPP, 3GPP2 and ETSI TISPAN by requiring support of networks other than GPRS, such as Wireless LAN, CDMA2000 and fixed line.
To ease the integration with the Internet, IMS uses IETF protocols wherever possible, e.g. Session Initiation Protocol (SIP). According to the 3GPP, IMS is not intended to standardize applications but rather to aid the access of multimedia and voice applications from wireless and wireline terminals, i.e. create a form of fixed-mobile convergence (FMC). This is done by having a horizontal control layer that isolates the access network from the service layer. From a logical architecture perspective, services need not have their own control functions, as the control layer is a common horizontal layer. However in implementation this does not necessarily map into greater reduced cost and complexity.
IP
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.
IP Centrex
Using IP-based network to provide centrex services such as call hold, call transfer, last number look-up and redial, call forward, three-way calling.
IP Clearinghouse
An organization that provides clearing and settlement services in IP Telephony.
IP Fragmentation
IP datagrams to be fragmented into pieces small enough to pass over a link with a smaller MTU than the original datagram size.
IP Phone
A VoIP phone has an Ethernet port (RJ-45) instead of a regular phone jack (RJ-11), and connects directly to a broadband Internet modem instead of to the phone line in your house. You can make and receive calls just like normal.
IP Precedence
see Type of Service.
IP Telephony
(Also called Voice over IP) Technology that allows voice phone calls to be made over the Internet or other Packet Switched Networks using traditional voice equipment such as handsets and telephony exchanges and voice gateways VoIP implementations are based on the standards, such as H.323 and SIP.
IPDC
IP Device Control (protocol).
IPDR
Internet Protocol Detail Record (protocol).
ISDN
Integrated Services Digital Network is a set of communications standards for simultaneous digital transmission of voice, video, data, and other network services over the traditional circuits of the public switched telephone network. It was first defined in 1988.
ISP
Internet Service Provider.
ITSP
Internet Telephony Service Provider.
ITU (International Telecommunications Union)
An organization established by the United Nations to set telecommunications standards, allocate frequencies for various uses, and sponsor trade shows every four years.
ITU-T
ITU standards for telecommunications.
IVR
In computer telephony, IVR (Interactive Voice Response) is a horizontal application wherein computer-based information is accessed over the phone by using a telephone instead of a computer. An IVR platform uses computer telephony components to translate callers’ touch-tones or voice commands into computer queries after the callers listen to 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 converted into voice commands that are spoken over the phone to the caller.
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Jingle
Jingle is an extension to the Extensible Messaging and Presence Protocol (XMPP) which adds peer-to-peer (P2P) session control (signaling) for multimedia interactions such as in Voice over IP (VoIP) or videoconferencing communications.
Jitter
The variation in the amount of Latency among Packets being received
Kamailio
Kamailio (formerly known as OpenSER) is an open source SIP Server released under GPL, able to handle thousands of call setups per second.
Features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video); IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra; XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.
Kilo-Bits Per Second (Kbps)
A measure of the number of one thousand bits transferred over a 1 second period.
Kilo-Hertz (KHz)
A measurement unit of the number of one thousand cycles per second of a waveform.
Lag
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
(Also called Delay) The amount of time it takes a Packet to travel from source to destination. Together, Latency and Bandwidth define the speed and capacity of a network.
LDCELP
Low-Delay Code Excited Linear Prediction.
LGPL
The GNU Lesser General Public License (formerly the GNU Library General Public License) or LGPL is a free software license published by the Free Software Foundation (FSF). It was designed as a compromise between the strong-copyleft GNU General Public License or GPL and permissive licenses such as the BSD licenses and the MIT License.
LinPhone
LinPhone is an open source video SIP phone for desktop and mobile. Runs on Windows, Linux, Mac OS X, Android, and iPhone. Supported audio codecs: Speex, G.711, GSM codec, and iLBC; video codecs: H.263, MPEG-4, Theora and H.264. Licensed under the GNU General Public License (GPL)
Load Balancing
Distribution of calls among terminating Gateways based on the Priorities and Weights assigned by the Buyer.
Local Area Network (LAN)
A group of computers and associated devices that share a common communications line or wireless link and typically share the resources of a single processor or server within a small geographic area (for example, within an office building).
Log
The log element allows an application to generate a logging or debug message, which a developer can use to help in application development or post-execution analysis of application performance.
Login ID
A string of digits identifying a Hostedswitch® Registered User. Together with the Password, the Login ID is used to authorize a user's access to the HostedSwitch® trading floor. The Login ID and Password are automatically e-mailed to a potential Hostedswitch® Member after filling out a Registration.
Lossless Codec
Media is compressed without any loss in quality.
LPC
Linear-Predictive Coding is a tool used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. It is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good quality speech at a low bit rate and provides extremely accurate estimates of speech parameters.
LPCP
Lightweight Phone Control Protocol.
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Mapping
The process of identifying all related data fields or data streams and putting them in an easily identifiable context. For example, IP mapping enables users to pinpoint the geographical location of any computing device on the Internet.
MCML PPP
Multi-Class Multilink Point-to-Point Protocol.
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.
Media Gateway (MG)
A translation unit between disparate telecommunications networks.
Media Gateway Controller (MGC)
A system used in MGCP/H.248/Megaco VoIP telephony architectures to control a number of Media Gateways.
Megaco
A IETF VoIP signaling protocol, same as H.248 of ITU-T.
MGCP (Media Gateway Control Protocol)
A protocol complementary to H.323 and SIP, designed to control media gateways from external call control elements in decomposed gateway architectures. Working in conjunction with the Gateway Location Protocol (GLP), MGCP enables a caller with a PSTN phone number to locate the destination device and establish a session. It provides the gateway-to-gateway interface for the Session Initialization Protocol (SIP). MGCP is meant to simplify standards for the new Voice over Packet technology by eliminating the need for complex, processor-intense IP Telephony devices, thus simplifying and lowering the cost of these terminals.
Minimum Duration
The minimum billed call duration up to which all shorter calls are rounded in seconds.
Minute
One (1) minute of communication resulting from a connection between a calling number and a called number.
MIT License
The MIT License is a free software license originating at the Massachusetts Institute of Technology (MIT). It is a permissive license, meaning that it permits reuse within proprietary software provided all copies of the licensed software include a copy of the MIT License terms. Such proprietary software retains its proprietary nature even though it incorporates software under the MIT License. The license is also GPL-compatible, meaning that the GPL permits combination and redistribution with software that uses the MIT License.
Modulation
Carrying information on a signal by varying one or more of the signal's basic characteristics -- frequency, amplitude and phase.
MPL
The Mozilla Public License (MPL) is a free and open source software license. The MPL is characterized as a hybridization of the modified BSD license and GNU General Public License.
MP-MLQ
Multi-Pulse, Multi-Level Quantization.
MTU
Maximum Transmission Unit.
MySQL
MySQL is the world's most used relational database management system (RDBMS) that runs as a server providing multi-user access to a number of databases. The MySQL development project has made its source code available under the terms of the GNU General Public License, as well as under a variety of proprietary agreements
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NANP
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.
NAT
Network Address Translation is a technique in which the source and/or destination addresses of IP packets are rewritten as they pass through a router or firewall. It is most commonly used to enable multiple hosts on a private network to access the Internet using a single public IP address.
NCS
Network-Based Call Signaling.
NGN
Next Generation Network.
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OpenSER
OpenSER is a Session Initiation Protocol (SIP) proxy server, call router, and user agent registration server used in Voice over Internet Protocol and instant messaging applications. Currently known as Kamailio.
OpenSIPS
OpenSIPS (Open SIP Server) is an open source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS unifies voice, video, IM and presence services in a highly efficient way.
Packet
In data communication, the basic logical unit of information transfered.
Packet Framing
A packet consists of two kinds of data: control information and user data (also known as payload). The control information provides data the network needs to deliver the user data, for example: source and destination addresses, error detection codes like checksums, and sequencing information. Typically, control information is found in packet headers and trailers, with user data in between.
Packet Loss
The term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latencyor on account of overloading of switches or routers that are unable to process or route all the incoming data.
Packet Switched Networks
These are networks that break messages into small digital packets, stamp each packet with the destination IP address, and route them across different channels to their destination where they are reassembled in their proper sequence. This is done to avoid network congestion and speed up data movement from multiple sources.
PCM
Pulse Code Modulation.
PDD
Post Dial Delay - 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.
Peer-to-Peer (P2P)
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.
PJSIP
Open source SIP stack and media stack for presence, IM/instant messaging, and multimedia communication. Designed to be very small in footprint, have high performance, and very flexible. PJSIP consists of multiple levels of APIs, which each of them layered on top of another.
Point-to-Point Protocol (PPP)
Point-to-point protocol (PPP) is a connection oriented protocol that is established between two communication devices that encapsulates data packets (such as Internet packets) for transfer between two communication points. PPP allows end users (end points) to setup a logical connection and transfer data between communication points regardless of the underlying physical connection (such as Ethernet, ATM, or ISDN).
PostgreSQL
PostgreSQL, often simply Postgres, is an object-relational database management system (ORDBMS) available for many platforms including Linux, FreeBSD, Solaris, Microsoft Windows and Mac OS X. It is released under the PostgreSQL License, which is an MIT-style license, and is thus free and open source software.
POTS
POTS is the short form of plain old telephone service. It transmits voice as analog data on communication lines that are much slower when compared to today’s ISDN or FDDI lines. However, not long ago POTS, which is also known as the public switched telephone network, was the standard telephone system across the world.
PQ-CBWFQ
Priority Queuing - Class-Based Weighted Fair Queuing.
PRI
Primary Rate Interface, a type of ISDN interface.
Priority
A number (1-10) assigned to an ordered (purchased) Contract or Route. If several Contracts/Routes for the same destination are ordered, Hostedswitch® will first attempt to terminate a call to the Contracts/Routes with the highest priority (1 is the highest and 10 is the lowest). If for some reason a connection cannot be established, Contracts/Routes with lower priority will be tried according to their respective priorities. See also: Weight.
Processor Drain
This is a term used to indicate a drop in the quality of VoIP phone service when a user opens several applications on his computer simultaneously.
Propogation Delay
The time required for a signal to travel from one point to another.
Protocol
In networking, the specification of a set of rules for a particular type of communication. The term also refers to the software that implements the protocol.
Proxy Server
Performs routing of a session invitations according to invitee's current location, authentication, accounting, etc.
PSTN
The Public Switched Telephone Network (PSTN) is the network of the world's public circuit-switched telephone networks. It consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all inter-connected by switching centers, thus allowing any telephone in the world to communicate with any other.
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Q.931
ISDN connection control protocol, roughly comparable to TCP in the Internet protocol stack. Q.931 doesn't provide flow control or perform retransmission, because the underlying layers are assumed to be reliable and the circuit-oriented nature of ISDN allocates bandwidth in fixed increments of 64 Kbps. Q.931 does manage connection setup and breakdown. In H.323 scenario, this protocol is encapsulated in TCP and sent to port 1720.
QoS (Quality of Service)
Ability of a network element (e.g., an application, host, or router) to have some level of assurance that its traffic and service requirements can be satisfied.
QSIG
RADIUS
Remote Authentication Dial In User Service (RADIUS) is a networking protocol that provides centralized Authentication, Authorization, and Accounting (AAA) management for computers to connect and use a network service.
Rating (Call Rating)
Rating is the activity of determining the cost of a particular call. The rating process involves converting call-related data into a monetary-equivalent value. Rating systems typically use some or all of the following types of data about a call:
- Time property of the call (day of week, date, time of day)
- Amount of usage (Duration of call, amount of data, number of messages, number of songs)
- Destination of the call (land line, overseas, etc.)
- Origin of call/ Location of the caller (for mobile networks)
- Premium charges (third party charges for premium content, cost of physical items such as movie tickets)
RAS (Registration, Admission, Status)
A management protocol between terminals and Gatekeepers.
Redirect Server
Receives a request and sends back a reply containing a list of the current location of a particular user.
Registered User
A person/company that has registered at Hostedswitch® Web Site and received an Account and ID/Password.
Registrar Server
Accepts REGISTER requests and places the information it receives in those requests into the location service for the domain in handles.
REST
REST is a style of software architecture that systems need in order to process and execute various software elements.
Route
A set of parameters predefined by HostedSwitch®to facilitate routing of traffic between the Gateways/Gatekeepers controlled by anHostedSwitch® Member either via ownership or via a partnership with the owner. Along with specifying other parameters, an Hostedswitch® Member using the Gatekeeping Service assigns to a Route values specifying the details of both originating and terminating Gateways/Gatekeepers.
Router
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.
RTP/RTCP (Real-time Transport Protocol/Real-time Control Protocol)
An IETF specification for audio and video signal management. RTP is used to send encoded voice in UDP packets. RTCP is used to send statistical and control information for a VoIP channel, such as the number of bytes sent, commands to enable/disable echo suppression, etc.
RTSP
Real Time Streaming Protocol
R-Factor
R-Factor is a VoIP transmission quality rating, with a typical range of 50-100. An R-Factor score is derived from multiple VoIP metrics, including latency, jitter, and loss.
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Sampling Rate
The number of samples per second taken from a continuous (analog) signal to make a discrete(digital) signal.
Sampling
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.
SAP
Session Announcement Protocol.
SBC (Session Border Controller)
SBC is an equipment deployed in Voice over Internet Protocol (VoIP) networks to exert control over the signaling and the media streams involved in setting up, conducting, and releasing telephone calls or other media communications.
SCTP
Stream Control Transmission Protocol.
SDP
Session Description Protocol.
SEMS
SEMS, SIP Express Media Server, is a free, high performance, extensible media and application server for SIP based VoIP services.
Service Provider
A service provider is a business entity that provides a communication, storage or processing service for a fee. Some of the service providers in the digital world are the Internet service provider (ISP), application service provider (ASP), storage service provider, mobile phone service provider, web hosting provider, and of course, VoIP service provider.
SGCP
Simple Gateway Control Protocol.
Signaling
The exchange of information between points in the network that sets up, controls, and terminates each telephone call.
SIGTRAN
A family of protocols that provides reliable datagram service and user layer adaptations for SS7 and ISDN communications protocols.
SILK
SILK is an audio compression format and audio codec used by Skype. SILK is a replacement for the SVOPC codec.
SIMPLE
SIMPLE (Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions) is an instant messaging (IM) and presence protocol suite based on Session Initiation Protocol (SIP) managed by the IETF. SIMPLE is an open standard.
Simple Network Management Protocol (SNMP)
Simple network management protocol, or SNMP, is an Internet protocol that is used for managing devices on IP networks.
SIP (Session Initiation Protocol)
An application-layer control protocol, a Signaling protocol for IP Telephony. SIP can establish sessions for features such as audio/videoconferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP Telephony services with Web, e-mail, and chat services. In addition to user authentication, redirect and registration services, SIP Server supports traditional telephony features such as personal mobility, time-of-day routing and call forwarding based on the geographical location of the person being called.
sipX
sipXecs (Enterprise Communications Server) is an open source IP Telephony server. Its main feature is a software implementation of the Session Initiation Protocol (SIP), which makes it an IP based communications system (IP PBX).
SOAP
SOAP, or simple object access protocol, is an Internet protocol that enables the exchange of web services and information via computer networks.
Softphone
This is a software application that is installed in the user’s PC. It uses the Voice over IP technology to route voice calls over the net and provides several value added features, such as call forwarding, conference calling, and integration with applications such as Outlook for automatic dialing The audio is provided through a microphone and speakers plugged into the sound card. The only limitation of a Softphone is that the phone call has to made through a PC. Many softphone are free VoIP software downloads..
Softswitch
(Also called a Proxy Gatekeeper, Call Server, Call Agent, Media Gateway Controller, or Switch Controller) Software used to bridge a public switched telephone network and voice over Internet by separating the call control functions of a phone call from the media gateway (transport layer). Softswitch performs call control functions such as protocol conversion, authorization, accounting and administration operations.
Softswitching Fee
The fee paid by Hostedswitch® Members to HostedSwitch® for the SoftSwitching Service. For pricing details please refer to the corresponding section of this site.
SoftSwitching Service
Softswitch service allows Hostedswitch® Members to bill, route and monitor IP Telephony traffic between their gateways and the gateways of their partners. For more information on this please refer to the corresponding section of this site.
Speech Power
The measure of the strength of a received voice signal.
Speex
Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Speex is available under the revised BSD license.
SRTP
Secure Real-time Transport Protocol.
SS7
Signaling System number 7.
STUN
Simple Traversal of UDP through NATs (Network Address Translation) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as 'Session Traversal Utilities for NAT'.
SVOPC
SVOPC (Sinusoidal Voice Over Pvacket Coder) is a compression method for audio which is used by VOIP applications. It is a lossy speech compression codec designed specifically towards communication channels suffering from packet loss. It uses more bandwidth than best bandwidth-optimised codecs, but it is packet loss resistant instead.
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T1
A wide-area digital transmission scheme (North American): 1,544 Mbits/s; 24 channels, 64 Kbps each.
T.120
ITU-T specifcation for Real-time Conferencing.
T.38
ITU-T specification for Facsimile over IP.
TCP (Transmission Control Protocol)
Connection-oriented transport layer protocol that provides reliable full- duplex data transmission. TCP is part of the TCP/IP protocol stack.
Theora
Theora is a free lossy video compression format. It is developed by the Xiph.Org Foundation and distributed without licensing fees alongside their other free and open media projects, including the Vorbis audio format and the Ogg container.
The SIP School
The SIP School provides first class education on SIP and VoIP along with industry recognized Certifications.
TLS (Transport Layer Security)
Transport Layer Security (TLS) is cryptographic protocol that provide communication security over the Internet. TLS encrypts the segments of network connections above the Transport Layer, using asymmetric cryptography for key exchange, symmetric encryption for privacy, and message authentication codes for message integrity.
ToS (Type of Service)
An 8-bit field in the IP datagram header that identifies the relative priority of one packet over another. Networking devices use this field to prioritize packets appropriately and place them in different queues if necessary.
Traffic Shaping
To control network traffic in order to optimize or guarantee performance, low latency, and/or bandwidth.
Trunk
In a communications network, a trunk is a transmission channel between two points that acts as either a switching centers or as a node. Possible examples are SIP trunks or H.323 trunks.
Twinkle
Twinkle is a softphone for voice over IP and instant messaging communcations using the SIP protocol. One can use it for direct IP Phone communication or in a network using a SIP proxy to route one's calls and messages.
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U-law
U-law is a lossy method of audio encoding that will compress 16-bit linear PCM audio samples into 8-bit samples thereby reducing bitrate by 50%. Like its encoding cousin, a-law, u-law assumes that the audio stream contains predominantly voice data (as opposed to, say, music data) which has a low dynamic range. It then uses a pseudo-logarithmic algorithm to compress the data, favoring detail for samples in the “middle” of the range and effectively glossing over samples at the extremes.
UDP (User Datagram Protocol)
A connectionless transport layer protocol in the TCP/IP protocol stack. UDP is a simple protocol that exchanges datagrams without acknowledgment or guaranteed delivery. Error processing and retransmission must be handled by other protocols.
Unified Messaging (UM)
The integration of different streams of messages (e-mail, Fax, voice, video, etc.) into a single in-box, accessible from a variety of different devices.
User Agents
A software program installed in a user’s terminal or an IP Phone to initiate and terminate phone calls.
Virtual Number
A virtual number is a telephone number without a directly associated telephone line. Forward-Phone-Number.com is one of the leading providers of virtual phone numbers around the globe.
VLAN
A virtual local area network, virtual LAN or VLAN, is a group of hosts with a common set of requirements, which communicate as if they were attached to the same broadcast domain, regardless of their physical location.
Voice over IP (VOIP)
VoIP or Voice over IP is the technology that is used to transmit voice over the Internet. The voice is first converted into digital data which is then organized into small packets. These packets are stamped with the destination IP address and routed over the Internet. At the receiving end the digital data is reconverted into voice and fed into the user’s phone.
Voicemail
It is a telephone messaging system that digitizes the analog voice signals and stores them on disk or flash memory in a central computer. These messages can then be retrieved by users by logging on to the server or forwarded to another voice mailbox. Most voice mail systems have auto attendant capabilities, that is they can use prerecorded messages to route callers to the appropriate person or mailbox. Voicemail is usually a free feature in VoIP service plans.
VOIP PBX
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.
Vorbis
Vorbis is a free software / open source project headed by the Xiph.Org Foundation. The project produces an audio format specification and software implementation (codec) for lossy audio compression.
VP7
TrueMotion VP7 is a video codec developed as a successor to earlier efforts such as VP3, VP5 and TrueMotion VP6. It is a codec with both VFW and DirectShow support. It is claimed to have better compression than leading competitive codecs such as MPEG-4 AVC (H.264) and VC-1.
VP8
VP8 is an open video compression format.
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WebM
The WebM is a project supported by Mozilla, Opera, Adobe, Google and more than seventy other publishers and software and hardware vendors and dedicated to developing a high-quality, open video format for the web that is freely available to everyone.
WebRTC
WebRTC is a free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs.
Weight
A number (10-100) assigned to a Contract or Route when ordering the Contract/Route. If several Contracts/Routes for the same destination have the same Priority assigned, calls to the destination are distributed among the Contracts/Routes according to their relative Weights.
Wide Area Network (WAN)
A communications network serving geographically separate areas. A WAN can be established by linking together two or more metropolitan area networks, which enables data terminals in one city to access data resources in another city or country.
Wireless
References the transmission of information (data, voice etc) over electromagnetic waves rather than over a wire connection.
WRED
Weighted Random Early Drop/Detect.
X.25
Approved by the CCITT (now the ITU-T) in the 1970s as a popular standard for packet-switching networks. It defines standard physical layer, data link layer and network layers (layers 1 through 3) of the OSI Reference Model. It was developed to describe how data passes into and out of public data communications networks. X.25 networks are in use throughout the world.
X264
A GPL-licensed implementation of H.264 encoding standard.
Xiph.Org Foundation
Xiph.Org Foundation (formerly Xiphophorus company) is a non-profit organization that develops free software tools and multimedia formats. It focuses on the Ogg family of formats, the most successful of which has been Vorbis. Current development work is concentrating on Theora.
Yate
Yate, Yet Another Telephony Engine, is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messenging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.
ZRTP
An extension to RTP which describes a method of Diffie-Hellman key agreement for SRTP.
3GPP
The 3rd Generation Partnership Project (3GPP) is a collaboration between groups of telecommunications associations, known as the Organizational Partners. The initial scope of 3GPP was to make a globally applicable third-generation (3G) mobile phone system specification based on evolved Global System for Mobile Communications (GSM) specifications within the scope of the International Mobile Telecommunications-2000 project of the International Telecommunication Union (ITU). The scope was later enlarged to include the development and maintenance of:
- the Global System for Mobile Communications (GSM) including GSM evolved radio access technologies (e.g. General Packet Radio Service (GPRS) and Enhanced Data Rates for GSM Evolution (EDGE))
- an evolved third Generation and beyond Mobile System based on the evolved 3GPP core networks, and the radio access technologies supported by the Partners (i.e., UTRA both FDD and TDD modes)
- an evolved IP Multimedia Subsystem (IMS) developed in an access independent manner.
3CX
3CX Phone System for Windows is a software-based IP PBX that replaces a proprietary hardware PBX / PABX. 3CX’s IP PBX has been developed specifically for Microsoft Windows and is based on the SIP standard. See http://www.3cx.com/
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