Confused about whether Voice over IP The technology has been gaining steam in the tech press during the past There’s no denying that IP-based calling has a future. Virtually all the Take a May, 2004, study of small, medium, and large organizations in
business to go? You’re not the only one.
while federal lawmakers and regulators wrangle over who will have the
regulatory upper hand.
major research firms predict more VoIP growth to come.
North America by telecommunications research firm Infonetics. The results
products and services in 2004.
The technology has been gaining steam in the tech press during the past
There’s no denying that IP-based calling has a future. Virtually all the
Take a May, 2004, study of small, medium, and large organizations in
respondent headquarter sites.
and over half will by March 2006.
average of 61 percent of respondent IP LAN endpoints; digital phones using
IP adapters are a distant second choice.
most likely providers to capitalize on managed IP voice services, but a full
42 percent of respondents say they don’t know who they’ll use.
phone systems and services work just fine.
merits of their IP voice products and services and how they can help address
today’s business challenges.
Not to mention the wrangling under way between state and federal lawmakers
over who should have regulatory oversight of the burgeoning technology.
To address some of the questions, internetnews.com compiled some key information about VoIP into this special section.
1. At-a-Glance Growth
1. At-a-Glance Growth: The Telecommunications Industry Association (TIA) expects VoIP to continue growing along with the adoption of broadband. Shipments of IP PBX lines, which converge voice and data, tell the VoIP story: They are surging, and the residential market is catching on. In a survey of U.S. Internet users, research firm Ipsos-Insight identified a number of VoIP features that are enticing would-be subscribers.
2. Tips from the Experts (Page 2 of 3 pages)
What should enterprise customers be aware of as they deploy VoIP?
James Puchbauer, Director of Marketing, AltiGen Communications
I like to say, “Technology should never be the siren song with which you crash your business on the rocks.”
Don’t buy something that is a tech-sounding solution. Make sure the system you buy is a mature application to
solve business needs, so that it can do the things you want it to do like integrate with a database or provide
centralized call recording.
Make sure the technology not only does VoIP, but also has the voice resources you need — can handle maximum capacity.
These voice resources do much more than just voicemail: There is interactive voice response; prompts play
when someone is in the sales queue; on-demand recording; and many more different valuable features you need.
All of those capabilities require voice resources, and you often don’t realize how expensive they are until
John Dretler, Senior Vice President, AnchorPoint
Before deployment, you need to define your current voice and data infrastructure; understand the volume and
impact of voice traffic on your data network; quantify the cost and savings, ROI, from moving your traffic;
and put in place the process to measure the ROI as you move forward.
Matthias Machowinski, Market Analyst, Infonetics Research
There are many types of solutions out there. Which one suits their particular requirements,
i.e. managed services vs. in-house deployments, hybrid vs. pure IP PBXs? This is not a one size fits all market.
Scott Testa, Co-Founder and COO, Mindbridge Software
Today we have 40 percent to 50 percent traffic on VoIP. A year ago it was at 20 percent, and two years ago it was at zero.
We did it gradually. I would say to most large enterprises, if you can do a gradual rollout, that
would be the preferred way of doing it. Try first as a secondary line system then,
after six months if you are happy with it, make a primary line in limited use and roll it out slow.
R. Pierce Reid, Vice President of Marketing, Qovia
The most successful implementations are those that have started with an “island” of VoIP in a department
or division of an enterprise. Through these smaller pilots, the network can be built outwards to a corporation-wide
network. Companies and entities that have tried “forklift” change outs of their phones and implementation of VoIP
have had Morse challenges.
The second thing to keep in mind is that a VoIP network requires care and feeding to run its best. In the same
way that an IT team has to monitor and manage their data network, they will need to manage their VoIP network.
But at the same time, a packet is not a packet. VoIP is much more critical in terms of timing and other factors.
So the same data NMS tools that worked for a data network don’t translate easily to VoIP. You need tools that
are designed for VoIP to manage VoIP networks.
Finally, VoIP network management is about more than QoS. It’s
about reliability, E911, security and other factors, as well. Remember, if you lose just a single 9 off your 99.999
percent reliability, that’s 40 hours of dial tone you lose each year. Which 40 hours would you like to lose?
When is it a good time to lose 40 hours of dial tone if you are a hospital or a sheriff’s department?
Here is a table that rounds up some VoIP regulatory issues.
2. Tips from the Experts/Regulatory Round-up
3. Glossary of VoIP Terms (Source: IPCB.net)(Page 3 of 3 pages)
A Contract that has been ordered.
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.
ANI [Automatic Number Identification]
A telephone function that transmits the billing number of the incoming call
(e.g. caller ID).
ANSI [American National Standards Institute]
American standardization body and member of the ISO [International Organization for Standardization].
A non-profit making, government-independent organization, it is known
for interface recommendations and standardization of programming languages.
AS [Autonomous System]
A group of networks under mutual administration that share the same routing methodology. 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 AS’s.
ASP [Application Service Provider]
An independent, third-party provider of software-based services delivered to customers across a WAN.
ATM [Asynchronous Transfer Mode]
Technology for switched, connection-oriented transmission of voice, data and video.
It makes high-speed dedicated connections possible between a theoretically unlimited number of network users and
also to servers. As a switching system (“Cell Relay”) it is to be used in broadband ISDN (B-ISDN) and also in the
Switched Multimegabit Data Service (SMDS networks). ATM is also becoming increasingly popular in the LAN area in
the form of ATM-LAN emulations. ATM is based on high-speed cell switching (packets of fixed size: 48+5 bytes)
that makes it possible to vary bit rates (according to requirements). In connection with ATM one speaks of message
blocks or message cells rather than message packets.
A very high-speed network spanning the world from one major metropolitan area to another. Such networks are
typically provided by national ISPs. Local ISPs connect to the backbone in order
to transport data.
Bad Frame Interpolation
Interpolates lost/corrupted packets by using the previously received voice frames. It
increases voice quality by making the voice transmission more robust in bursty error environments.
See Net Termination Balance.
The maximum data-carrying capacity of a transmission link. For networks, bandwidth is usually expressed in bits
per second (bps).
Federal Data Protection Act
A call duration measurement unit expressed in seconds.
BLI [Busy Lamp Indicator]
A light or LED on a telephone that shows which line is in use.
Descriptive term for evolving digital technology that provides consumers a single switch facility offering
integrated access to voice, high-speed data service, video demand services, and interactive delivery services.
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.
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).
Used at anywhere from 1:1 to 12:1 ratios in VoIP applications to consume less bandwidth and leave
more for data or other voice/fax communications. The voice quality may decrease with increased compression ratios.
The situation in which the traffic present on the network exceeds available network bandwidth/capacity.
Mode of communication in which a connection must be established between the transmitter and receiver before transmission
of user data. This can be done by switching a circuit or by setting up a logical channel. A well-known connection-oriented
protocol is TCP. Connection-oriented is the opposite of connectionless.
Mode of communication in which a connection (circuit or logical channel) does not need to be set up for data
transmission between the transmitter and receiver. It is the underlying protocol for packet-switched transmission.
The individual data packets can go from the transmitter to the receiver via different paths. A well-known
connectionless protocol is UDP.
A set of parameters that an IPCB.net Member using the VoIP Termination Service establishes in order to receive traffic
from and provide termination services to other IPCB.net Members. Contract details include the requested price per minute
(Tariff), Grace Period, Minimum Call Duration, Billing Increment, and one or more registered Gateways/Gatekeepers that
will be terminating calls sent by an IPCB.net Member who has ordered this Contract.
CSMA/CD [Carrier Sense Multiple Access/Collision Detection]
This is the access procedure to the Ethernet in which the
participating stations physically monitor the traffic on the line. If no transmission is taking place at the time, the
particular station can transmit. If two stations attempt to transmit simultaneously, this causes a collision, which is
detected by all participating stations. After a random time interval, the stations that collided attempt to transmit
again. If another collision occurs, the time intervals from which the random waiting time is selected are increased
step by step. Networks using the CSMA/CD procedure are simple to implement but do not have deterministic transmission
characteristics. The CSMA/CD method is internationally standardized in IEEE 802.3 and ISO 8802.3.
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.
Process when the originating router tries to establish a 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.
DID [Direct Inward Dialing]
The ability to make a telephone call directly into an internal extension without having to
go through the operator.
DiffServ [Differentiated Services]
A quality of service protocol that prioritizes IP voice and data traffic to
help preserve voice quality even when network traffic is heavy.
DNIS [Dialed Number Identification Service]
A telephone function that sends the dialed telephone number to the answering
DTMF [Dual-Tone Multifrequency]
The type of audio signals generated when you press the buttons on a touch-tone telephone.
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.
E&M [Ear and Mouth]
The interface on a VoIP device that allows it to be connected to analog PBX trunk ports (tie lines).
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.
A wide-area digital transmission scheme (European): 2,048 Mbits/s; 31 channels, 64 Kbps each.
SIP or H.323 terminal or Gateway. An endpoint can call and be called. It generates and terminates the information stream.
A system designed to prevent unauthorized access to or from a private network. Firewalls can be implemented as
hardware or 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 the security criteria specified on the firewall.
Forward Error Correction
Increases voice quality by recovering lost or corrupted packets.
FXO [Foreign Exchange Office]
The interface on a VoIP device for connecting to an analog PBX extension.
FXS [Foreign Exchange Station]
The interface on a VoIP device for connecting directly to phones, faxes, and CO ports
on PBXs or key telephone systems.
An ITU-T double rate CELP codec (6.4/5.3 kbps, medium quality, high processor load).
An ITU-T ADPCM wave form codec (16/24/32/40 kbps, good quality, low processor load).
An ITU-T low delay CELP codec (16 kbps, medium quality, very high processor load).
An ITU-T ACELP codec (8 kbps, medium quality, high processor load).
A family of ITU standards for audio compression.
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.
GKTMP [Gatekeeper Transaction Message Protocol]
A proprietary Cisco protocol used for communication between the Cisco IOS Gatekeeper and external applications.
Protocols (RAS, RTP/RTCP, Q.931 call signaling) and message formats for H.323.
A protocol for capability negotiation, messages for opening and closing channels for media streams, etc.
(i.e. media signaling).
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.
Means to send a call back in the direction that it came from. For example, if a call cannot be
routed over IP to a gateway that is closer to the target telephone, the call typically is sent back out the local
zone, back the way from which it came.
Point at which a call transitions from H.323 to non-H.323, typically at a gateway.
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.
See Type of Service.
The transmission of voice and fax phone calls over data networks that use the IP. IP telephony
is the result of the transformation of the circuit-switched telephone network to a packet-based network that deploys
voice-compression algorithms and flexible and sophisticated transmission techniques, and delivers richer services
using only a fraction of traditional digital telephonys usual bandwidth. Compare with VoIP.
Internet Telephony Service Provider.
ITU standards for telecommunications.
The variation in the amount of Latency among Packets being received.
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. (Also called “Delay”)
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.
The basic logical unit of information transfered.
PBX [Private Branch eXchange]
An in-house telephone switching system that interconnects telephone extensions to each
other, as well as to the outside telephone network.
PRI [Primary Rate Interface]
An ISDN service that provides 23 64-Kbps B (Bearer) channels and one 64-Kbps D (Data)
channel (23 B and D).
Public Switched Telephone Network.
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]
Measure of performance for a transmission system that reflects its transmission quality and
service availability. Standards based-QOS for VoIP usually involves the implementation of Ethernet standards 802.1p
and 802.1q at layer 2 across an Ethernet. At layer 3, the IP standard DiffServ defines bits settings in the TOS
(type of service) in the IP header, which will identify packets as being associated with a specific service.
QSIG [Q (point of the ISDN model) Signaling]
Common channel signaling protocol based on ISDN Q.931
standards and used by many digital PBXs.
RAS [Registration, Admission, Status]
A management protocol between terminals and gatekeepers.
RSVP [Resource Reservation Protocol]
Protocol that supports the reservation of resources across an IP network. Applications
running on IP end systems can use RSVP to indicate to other nodes the nature
(bandwidth, jitter, maximum burst, and so on) of the packet streams they want to receive. RSVP depends on IPv6.
(Also known as Resource Reservation Setup Protocol.)
RTP [Real-Time Transport Protocol]
Commonly used with IP networks. RTP is designed to provide end-to-end network transport
functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or
unicast network services. RTP provides such services as payload type identification, sequence numbering, timestamping,
and delivery monitoring to real-time applications.
SIP [Session Initiation Protocol]
An application-layer control protocol; a signaling protocol for Internet 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.
(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.
1.544-Mbps point-to-point dedicated digital circuit provided by the telephone companies consisting of 24 channels.
TAPI [Telephony API]
A programming interface that allows Windows client applications to access voice services on a server.
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.
A communications channel between two points, typically referring to large-bandwidth telephone channels between
switching centers that handle many simultaneous voice and data signals.
VoIP [Voice over IP]
The capability to carry normal telephony-style voice over an IP-based Internet with POTS-like
functionality, reliability, and voice quality. VoIP enables a router to carry voice traffic
(for example, telephone calls and faxes) over an IP network. In VoIP, the DSP segments the voice signal into frames,
which then are coupled in groups of two and stored in voice packets. These voice packets are transported using IP in
compliance with ITU-T specification H.323.
VPDN [Virtual Private Dial-up Network]
A network that extends
remote access to a private network using a shared infrastructure. VPDNs use Layer 2 tunnel technologies
(L2F, L2TP, and PPTP) to extend the Layer 2 and higher parts of the network connection from a remote user across
an ISP network to a private network. VPDNs are a cost-effective method of establishing a long-distance, point-to-point
connection between remote dial users and a private network. (Also known as virtual private dial network.)
VPN [Virtual Private Network]
Enables IP traffic to travel securely over a public TCP/IP network by encrypting all
traffic from one network to another. A VPN uses “tunneling” to encrypt all information at the IP level.
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.
3. Glossary of VoIP Terms