VoIP Internet
Phone Service

Voice-over-IP service from VoIPsvc.net: nationwide and international phone calls to traditional phones at local rates and free calls within the VoIP network.

Glossary of Terms


Average Call Duration.


An open source communications platform, Asterisk® is a complete IP PBX in software. It runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX.


Trixbox (formerly Asterisk@Home), the asterisk-based solution, enables the home or small business user to quickly set up a VOIP Asterisk-based PBX. A web GUI makes configuration and operation easy (http://www.trixbox.com).


An analog telephony adapter, or analog telephone adapter (ATA), is a device used to connect one or more standard analog telephones to a digital and/or non-standard telephone system such as a Voice-over-IP based network.

An ATA usually takes the form of a small box with a power adapter, one Ethernet port, and one or more FXS telephone ports. Users can plug one or more standard analog telephone devices into the ATA and the analog device(s) will operate, usually transparently, on a VoIP network.


Caller ID (caller identification or CID and, more properly, calling number identification - CNID) is a telephony service that transmits the caller's telephone number to the called party's telephone equipment during the ringing signal or when the call is being set up but before the call is answered. Where available, Caller ID can also provide a name associated with the calling telephone number. The information made available to the called party is visible on a small liquid crystal display imbedded on the telephone, or on a separate unit which is connected to the telephone.

Call origination

Call Origination, also known as voice origination, refers to the collecting of the calls initiated by a calling party on a telephone exchange of PSTN, and handing off the calls to a VoIP endpoint or to another exchange or telephone company for completion to a called party.


Telephone exchanges generate so called Call Detail Records (CDRs) which contain detailed information about calls originating from, terminating at or passing through the exchange. Not surprisingly CDRs are used for billing.


Cisco Systems, Inc. (NASDAQ: CSCO, SEHK: 4333) is a global company headquartered in San Jose, California, USA, that designs and sells networking and communications technology and services under four brands: Cisco, Linksys, WebEx, and Scientific Atlanta. Initially, Cisco manufactured only enterprise multi-protocol routers, but today Cisco's products can be found everywhere from the living room to the enterprise to service provider networks. Cisco's vision is "Changing the Way We Live, Work, Play and Learn." Cisco's current tagline is "Welcome to the human network."


A CNAM database contains calling party names to be used when identifying a calling party. We offer optional CNAM service on our US and Canadian DIDs as well as on our toll free DIDs.


Voice transmission is analogical, whereas the data network is digital. The process to sample analogical waves into digital information is made by an encoder-decoder (CODEC). There are many standards to sample an analogical voice signal into a digital one. The process is often quite complex. Most of the conversions use pulse code modulation (PCM) or variations. We support the following codecs:

Codec Bit Rate Nominal Ethernet Bandwidth (One Direction)
G.711u 64 Kbps 87.2 Kbps
G.729a 8 Kbps 31.2 Kbps
GSM 13 Kbps 29.2 Kbps


The comma-separated values (or CSV, also known as a comma-separated list or comma-separated variables) file format is a file type that stores tabular data. The format dates back to the early days of business computing. For this reason, CSV files are common on all computer platforms.

CSV is one implementation of a delimited text file, which uses a comma to separate values. However, CSV differs from other delimiter separated file formats in using a " (double quote) character around fields that contain reserved characters (such as commas or newlines). Most other delimiter formats either use an escape character such as a backslash, or have no support for reserved characters.


Direct Inward Dialing. "DID" numbers have particular relevance for VoIP communications. In order for people connected to the traditional PSTN network to call people connected to VoIP networks, DID numbers from the PSTN network are obtained by the administrators of the VoIP network, and assigned to a gateway in the VoIP network. The gateway will then route calls incoming from the PSTN across the IP network to the appropriate VoIP user. Similarly, calls originating in the VoIP network will appear to users on the PSTN as originating from one of the assigned DID numbers, if the user setup his callerid accordingly.


E.164 is an ITU-T recommendation which defines the international public telecommunication numbering plan used in the PSTN and some other data networks. It also defines the format of telephone numbers. E.164 numbers can have a maximum of 15 digits and are usually written with a + prefix. To actually dial such numbers from a normal fixed line phone the appropriate international call prefix must be used.


FreePBX is the most powerful GUI (Web Based) configuration tool for Asterisk. It provides everything that a standard legacy phone system can, plus a huge amount of new features. All documentation and information is avalable from http://www.freepbx.org. FreePBX is included in the trixbox distribution.


G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. G.711 represents logarithmic pulse-code modulation (PCM) samples for signals of voice frequencies, sampled at the rate of 8000 samples/second.

There are two main algorithms defined in the standard, the µ-law algorithm (used in North America & Japan) and A-law algorithm (used in Europe and the rest of the world). Both are logarithmic, but A-law was specifically designed to be simpler for a computer to process. The standard also defines a sequence of repeating code values which defines the power level of 0 dB. The µ-law and A-law algorithms encode 14-bit and 13-bit signed linear PCM samples (respectively) to logarithmic 8-bit samples. Thus, the G.711 encoder will create a 64 kbit/s bitstream for a signal sampled at 8 kHz.


G.723 is a ITU-T standard wideband speech codec. This is an extension of Recommendation G.721 adaptive differential pulse code modulation to 24 and 40 kbit/s for digital circuit multiplication equipment application. Superseded by G.726, this standard is obsolete. Note that this is a completely different codec from G.723.1.


G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames. An algorithmic look-ahead of 7.5 ms duration means that total algorithmic delay is 37.5 ms. Note that this is a completely different codec from G.723. There are two bitrates at which G.723.1 can operate:

  • 6.3 kbit/s (using 24 byte frames) using a MPC-MLQ algorithm (MOS 3.9)
  • 5.3 kbit/s (using 20 byte frames) using an ACELP algorithm (MOS 3.62)

G.723.1 is mostly used in Voice over IP (VoIP) applications due to its low bandwidth requirement. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus some other method such as G.711 or out-of-band methods should be used to transport these signals. The complexity of the algorithm is below 16 MIPS. 2.2 kilobytes of RAM is needed for codebooks.


G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new 16 kbit/s rate. The four bit rates associated with G.726 are often referred to by the bit size of a sample, which are 2-bits, 3-bits, 4-bits, and 5-bits respectively.

The most commonly used mode is 32 kbit/s, since this is half the rate of G.711, thus increasing the usable network capacity by 100%. It is primarily used on international trunks in the phone network. It also is the standard codec used in DECT wireless phone systems and is used on some Canon cameras.


G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. Music or tones such as DTMF or fax tones cannot be transported reliably with this codec, and thus use G.711 or out-of-band methods to transport these signals.

G.729 is mostly used in Voice over IP (VoIP) applications for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and better speech quality respectively. Also very common is G.729a which is compatible with G.729, but requires less computation. This lower complexity is not free since speech quality is marginally worsened. G.729 is patented by Sipro in a number of countries. The use of G.729 may require a license fee and/or royalty fee.


H.323 is an umbrella recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. It is currently implemented by various Internet real-time applications such as NetMeeting and Ekiga (the latter using the OpenH323 implementation). It is a part of the H.32x series of protocols which also address communications over Integrated Services Digital Network (ISDN), Public switched telephone network (PSTN) or Signaling System 7 (SS7). H.323 is commonly used in Voice over IP (VoIP, Internet Telephony, or IP Telephony) and Internet Protocol (IP)-based videoconferencing. Its purpose is thus similar to that of the Session Initiation Protocol (SIP).


We refer to inbound as traffic we receive and that is directed to you, such as calls to your DID numbers.

Billing Increments

It's the way we calculate our rates in order to bill your calls. For example, if you call USA for 10 seconds, you will be charged for 12 seconds (2 x 6 seconds since this is a 6 seconds increment call) of a minute, not the whole minute.


Jitter is a typical problem of the connectionless networks or packet switched networks. Due to the information is divided into packets each packet can travel by a different path from the emitter to the receiver.

Jitter is technically the measure of the variability over time of the latency across a network. Real time communications (for example VoIP) usually have quality problems due to this effect. In general, it is a problem in slow-speed links or with congestion. It is hoped that the increase of QoS (quality of the service) mechanisms like priority buffers, bandwidth reservation or high-speed connections (Ethernet, E3/T3, SDH) can reduce jitter problem in the future although it will keep on being a problem for a long time.


Latency has the reputation of being the enemy of VoIP. It is also called lag. Latency is the time between the moment a voice packet is transmitted and the moment it reaches its destination. It of course leads to delayand finally to echo. It is caused by slow network links. This is what leads to echo.

There are two ways latency is measured: one direction and round trip. One direction latency is the time taken for the packet to travel one way from the source to the destination. Round-trip latency is the time taken for the packet to travel to and from the destination, back to the source. In fact, it is not the same packet that travels back, but an acknowledgement.

Latency is measured in milliseconds (ms) - thousandths of seconds.


IAX is the Inter-Asterisk eXchange protocol used by Asterisk, a dual licensed open source and commercial PBX server from Digium and other softswitches and PBXs. It is used to enable VoIP connections between servers, and between servers and clients that also use the IAX protocol. IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated almost universally in favor of IAX2.

NAT Transversal

NAT (Network Address Translation) is a technology most commonly used by firewalls and routers to allow multiple devices on a LAN with 'private' IP addresses to share a single public IP address. A private IP address is an address, which can only be addressed from within the LAN, but not from the Internet outside the LAN. In order to let a device with a private IP address communicate with other devices on the Internet, there needs to be a translation between private and public IP addresses at the point where the LAN connects to the Internet, that is within the firewall/router connecting the LAN to the Internet. Such a translation is commonly referred to as NAT (for Network Address Translation) and a router doing such translation is often called a NAT router or NAT firewall/router. Sometimes NAT is also called IP Masquerading. The passing of traffic through NAT is called NAT Traversal.

Packet Loss

VOIP is not tolerant of packet loss. Even 1% packet loss can "significantly degrade" a VOIP call using a G.711 codec and other more compressing codecs can tolerate even less packet loss. Cisco says: The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors. Ideally, there should be no packet loss for VoIP.


An IP PBX is a private branch exchange (telephone switching system within an enterprise) that switches calls between VoIP (voice over Internet Protocol or IP) users on local lines while allowing all users to share a certain number of external phone lines. The typical IP PBX can also switch calls between a VoIP user and a traditional telephone user, or between two traditional telephone users in the same way that a conventional PBX does. The abbreviation may appear in various texts as IP-PBX, IP/PBX, or IPPBX.

Predictive Dialer

A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers.

Prepaid Model

Prepaid refers to services paid for in advance. Examples include tolls, pay as you go cell phones, and stored-value cards such as gift cards and preloaded credit cards. Prepaid options can have substantial cost reductions over postpaid counterparts because they allow customers to monitor and budget usage in advance.


The public switched telephone network (PSTN) is the network of the world's public circuit-switched telephone networks, in much the same way that the Internet is the network of the world's public IP-based packet-switched networks. Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital, and now includes mobile as well as fixed telephones. It is sometimes referred to as the Plain Old Telephone Service (POTS).


SIP Express Router (SER) is a high-performance, configurable, free software SIP (cit. RFC 3261) server. It can act as SIP registrar, proxy or redirect server. SER features presence support, RADIUS/syslog accounting and authorization, XML-RPC-based remote control, etc. Web-based user provisioning, serweb, is available. SER's performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and a rapidly growing user population. SER can be configured for many scenarios including small-office use, enterprise PBX replacements and carrier services. SER is publicly available under the terms of the GNU General Public License.


"The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences." (cit. RFC 3261). It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 from the IETF SIP Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is widely used as a signaling protocol for Voice over IP, along with H.323 and others.

Soft Phone

A softphone is a software for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or with a USB phone.

Soft Switch

A softswitch is a central device in a telephone network that connects calls from one phoneline to another, entirely by means of software running on a computer system. This work was historically carried out by hardware, using physical switchboards to route the calls.


Call Termination, also known as voice termination, refers to the handing off or routing of calls from one telephone company, also known as a carrier or provider, to another telephone company. The terminating point is end point. The originating point is the party who initiates the call.

The term is highly used when referring to calls while using voip: a call initiated as a VoIP call is terminated using the PSTN. The opposite of call termination is call origination, where a call initiated from the PSTN is terminated using VoIP.


An asterisk-based solution, Trixbox, enables the home or small business user to quickly set up a VOIP Asterisk-based PBX. A web GUI makes configuration and operation easy (http://www.trixbox.com).

Sources: Asterisk, voip-info.org, whatis.com, Wikipedia.

Back to Previous Page