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This FAQ is edited by: KeysCapt See Profile, rjackson See Profile
It was last modified on 2010-09-24 18:28:13

1. What is VoIP?

What is VoIP?

VoIP stands for Voice over Internet Protocol.

As the term says VoIP is the transmission of voice communication through IP packets and, therefore, through the Internet. VoIP can use accelerating hardware to achieve this purpose and can also be used in a PC environment.

Webopedia defines it as: A category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls. For users who have free, or fixed-price Internet access, Internet telephony software essentially provides free telephone calls anywhere in the world. To date, however, Internet telephony does not offer the same quality of telephone service as direct telephone connections.

There are many Internet telephony applications available. Some, like CoolTalk and NetMeeting, come bundled with popular Web browsers. Others are stand-alone products. Internet telephony products are sometimes called IP telephony, Voice over the Internet (VOI) or Voice over IP (VOIP) products.

If you're wondering whether this is something that can just replace your standard home telephone service, the answer is that many are doing just that, and Vonage seems to be the popular choice. See the Vonage section of this FAQ, and check out users' comments about the service in the forum.

How can I submit an entry for this FAQ?

Any member can submit a suggested addition to this FAQ. To do so, click here and use the form to submit your entry. It's best if you list your question and include an answer, if possible, but you can submit the question by itself if you do not have an answer.

2. How does it work?

How does VoIP work?

Early on it was determined that it was possible to send a voice signal to a remote destination digitally, as well as via analog. To do that, we have to digitize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it.

This is basically the way VoIP works, sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service.

What do I need?

In order to use some of the basic services, (net2phone, IConnectHere), you can probably get away with just your PC, an internet connection, and your speakers/microphone. Most feel that using a headset is a much better solution.

In general, you need:

A high-speed internet connection. This can be a cable modem, or a super high-speed services such as ISDN, DSL or a T-1 link. The link you choose will depend upon the amount of telephone traffic you intend to use. A typical DSL link, for instance, is enough for eight simultaneous phone calls.

A VOIP box - about the size of a telephone modem. - your telephones just plug into the box. And your ordinary phone lines allow remote users to call in from outside.

An account with a VoIP provider. Your account can be configured to recognize your calls automatically, or you can require your users to enter their unique account numbers, issued by you.

Some examples of options in this area can be seen on IConnectHere's web site here.

These include headsets and handsets, both normal and USB, gateways, and IP phones.

3. What are the advantages?

Why would I be interested?

When you are using Public Switched Telephone (PSTN) line, you typically pay for the time you use: The more time you stay on the phone the more you'll pay. And you generally don't have the option of talking with more than one person at a time (or you can, but at increased cost).

With VoIP, you can talk all the time with any person you want (the requirement is that the other person has an internet connection), with no regard to distance, and you can talk with many people at the same time.

Not convinced? Consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos.

Why isn't everybody using it?

Like many new technologies, there have been some problems with VoIP. In order to be effective, voice data communication has to be a real time stream. (You wouldn't want to finish a sentence, then wait for many seconds before you could hear the other side answering.)

With normal internet protocols, you can have a network consisting of many routers (20 - 30 or more) which can result in a very high round trip time (RTT).

In addition to standard IP, VoIP uses the real-time protocol (RTP) to help ensure that packets get delivered in a timely way. Using public networks, it is currently difficult to guarantee Quality of Service (QoS). Better service is possible with private networks managed by an enterprise or by an Internet telephony service provider (ITSP).

As of this writing, VoIP is still plagued by lack of generally accepted interoperability standards.

3.6 User Reviews

A Vonage User Review

See this thread for one user's review of Vonage.

A Packet8 User Review

See this thread for one user's review of Packet8.

A user chooses between Vonage and Optimum Voice (OV)

In this thread a user recounts his experience with OV after making the choice between that service and Vonage.

4. Overview of VoIP Technology


A VoIP gateway can be loosely defined as a mechanism that takes circuit-switched voice from a traditional PBX, converts it to IP and transfers it across a LAN or WAN to another gateway where it is reconstituted back into a format that is understood by the receiving phone system.

Gateway functionality can be obtained through stand-alone boxes, modules or chassis cards for proprietary boxes; also expandable routers or software and expansion cards for Windows NT servers.

One of the front-runners in VoIP, Cisco is taking a modular approach with a voice-over-IP card that fits its routers. Cisco says all of its products can easily be equipped for voice and that voice packets can be guaranteed via quality-of-service (QoS) policy implementation on a Cisco-switched network.

Lucent, Nortel Networks and Siemens offer similar strategies for providing VoIP gateway capabilities in some form or another.

While gateways are the most popular VoIP products on the market -- available from at least 30 vendors -- the key point here is that you have voice packets running over IP. However, the packets are not running on the Internet, and you're not gaining any of the features and capabilities you get by converging voice and data networks.

You can find a list of gateway vendors Here


IP PBXs are great if you have the luxury of designing your system from the ground up. They are complete phone systems, usually with options that include many of the IP telephony applications, such as managing your phone from your desktop PC, multi-line call control and automatic call distribution.

IP PBXs are usually NT servers with telephony software and voice cards. Some disadvantages are the ability to scale the system and a dial tone that's dependent on NT, which doesn't offer the same uptime as a switched phone network.

Until recently IP PBXs have been targeted at offices with 100 users or less, but Alcatel recently announced a system that incorporates gateway and call processing in a single device and can accommodate up to 50,000 users. 3Com, Lucent and Cisco have all announced plans to provide the same type of product.

The beauty of an IP PBX is being able to create a distributed system. For example, allowing you to distribute your phone system throughout an IP network, so geographically separated phones with features such as direct dial, call forwarding, conferencing and voice mail provide the appearance of being connected directly to the local PBX.

Elements of a VoIP Network

VoIP services need to be able to connect to traditional circuit-switched voice networks.

The H.323 standard components are: H.323 terminals that are endpoints on a LAN, gateways that interface between the LAN and switched circuit network, a gatekeeper that performs admission control functions and other chores, and the MCU (Multipoint Control Unit) that offers conferences between three or more endpoints.

H.323 terminals are LAN-based end points for voice transmission. Common examples are a computer running Microsoft NetMeeting and an Ethernet-enabled telephone. All H.323 terminals support real-time, 2-way communications with other H.323 entities.

The gateway serves as the interface between the H.323 and non-H.323 network. On one side, it connects to the traditional voice world, and on another side to packet-based devices.

The gatekeeper is not a required device in an H.323 network, but if present, it must perform a set of functions. Gatekeepers provide address translation (routing) for devices in their zone. This could be, for instance, the translation between internal and external numbering systems. Another important function for gatekeepers is providing admission control, specifying what devices can call what numbers.

MCU's allow for conferencing functions between multiple terminals. They contain two parts:

A multipoint controller (MC) that handles the signalling and control messages necessary to setup and manage conferences, and a multipoint processor (MP) that accepts streams from endpoints, replicates them and forwards them to the correct participating endpoints.


In contrast to broadcast-type media transmission (RealAudio, for instance), latency in a two-way phone conversation can be very undesirable. 150ms is the specified maximum desired one-way latency to achieve high-quality voice. Voice users will notice round-trip delays that exceed 250ms. More than that, and callers start talking over each other. At anything beyond 500ms phone calls are no longer practical. It would be like asking a question and getting the answer after you've gone to lunch. As a reference, the typical delay when speaking through a geo-stationary satellite is 150-500ms.

Delay in data networks is not as critical. An additional delay of 200ms on an e-mail or web page will generally not be noticed, but voice callers on the same network would be hampered by such a delay.

Burglar Alarm Systems & VOIP

This is a hotly debated topic. The short answer: many alarm companies do not want to support VoIP, which can present considerable obstacles for those turning to VOIP. If you get past that point, you will find some interesting discussions of how reliable VoIP is for this type of communications in this forum thread: VoIP and Burglar Alarms. It seems that it can be done, but once you're past the technical hurdles you need to decide for yourself whether you want to trust the public internet for your alarm communications.

Additional information to help you make the decision to switch to VoIP can be found in this thread: So you wanna try Voip?

4.2 VoIP and FAX

Can I FAX with VoIP?

This question comes up on a regular basis. The answer is "maybe". Some providers allow it, some don't.

See this thread for some information. If your provider isn't mentioned there, do a forum search on "FAX" and you may turn up a thread where your provider is discussed.

5. Hardware and Software

What are some VoIP software/hardware?

How to reset a Cisco ATA186 to factory defaults

If you have an ATA186 that was previously used with a provider that password protected the configuration options, you will can clear the ATA186 of the password and reconfigure the device itself. This is good if you want to experiment with trying to make two connections from one ATA186 to another.

DO NOT perform this procedure if you are using the ATA186 with a currently connected provider. You WILL lose ALL configuration information for that provider, and will have to re-enter the information again in order to restore your service. I have included instructions that MAY work for re-enabling the ATA186 to use with Vonage, but keep in mind it will only work to restore your Vonage service.


Also, keep in mind if you use this procedure, and then restore your previous connection settings, the original settings from your provider will overwrite anything you may have changed, and the box will once again be password protected with the information that is required for your service.

The following procedure will reset the ATA to its default config:
a) Take the phone off hook.
b) The red button on the top of the ATA-186 will illuminate.
c) Press the illuminating red button on the ATA and dial 322873738#. (The numbers spell FACTRESET# on the telephone)
d) Voice prompt will ask you to dial * to save changes you have just made.
e) Press * on your phone's keypad.
f) Hang up the phone.


To get to the ATA186 web configuration page, go to »ipaddress/dev -- So if your router assigned your ATA186 an IP address of, you would go to »

To try and restore Vonage service, plug the following information in to the configuration page:

UseTFTP: 1
TFTP Url: Try either or
Encrypt Key: 1

Click Apply. Then wait about a minute. Check to see if your dialtone has returned.

Keep in mind, DO THIS AT YOUR OWN RISK. I strongly suggest you DO NOT try this if you have currently active service on your ATA186. Since I cancelled my Vonage line, I have no way of verifying if this procedure for restoring the service will work or not.

Hope this helps anyone that was curious as to how to do it.

From a post by RickNY

ATA 186 configuration guides

This link may provide some assistance in configuring and resetting the ATA-186:

Cisco ATA-186

Codec Information

Codec is short for code/decode.

MSDN defines a codec as:
A list of "instructions," known as an algorithm, which programs such as encoders and players use to compress and decompress data. In the case of digital media content, codecs are used to decrease the content's file size and bit rate, or the amount of data in kilobits per second that are required to render audio and video content. With smaller file sizes and lower bit rates, digital media content can be stored and streamed over a network more easily.

You may also see an error message in Windows Media Player (WMP) that says "Error downloading Codec." WMP tries to download the right codec if you try to play a video for which you don't have a codec installed on your system. If you are behind a firewall or router, WMP may not be able to connect to the Microsoft server to automatically grab the right one. You may be able to resolve this issue by temporarily disabling your firewall. WMP may also not be able to download the right codec if it cannot determine the internal format of the movie file.

One solution is to visit a site such as The Nimo Codec Homepage where you can download a huge file that will install a variety of different codecs on your system. Once you have done this, you should have no trouble playing movie files in the future.

Other useful sites for codecs include:
MS Codecs Download Package
Free Codecs

How to configure Sipura 2100 behind router

This is a quote from the Sipura 2100 FAQ:

I connect my SPA-2100 to another router which gives out 192.168.0.x IP address.

Please realize that the LAN IP address of SPA-2100 has an IP address of, and it will provide DHCP addresses on that subnet. If you have another router that this SPA connects to, be sure that the router does not assign the same subnet of 192.168.0. Otherwise it will cause a conflict with SPA-2100.

Solution change your router to give out IP addresses on a different subnet, i.e: 192.168.10.

Alternatively, you can change SPA-2100's LAN IP address also and its DHCP assignment. To connect to the SPA at this state: unplug the WAN connection, and reboot the SPA. Connect your PC to the LAN port, then to connect to » and make the above suggested change.

Submitted by keithpetersen

How do you configure Cisco CallManager Express to utilize a SIP VoIP provider?

Here is a config template to use to configure a Cisco router to access a SIP VoIP provider. Link to the full config description

voice translation-rule 2
rule 1 /614xxxxxxx/ /xxxxxxx/
voice translation-rule 7
rule 1 /^2../ /614xxxxxxx/
voice translation-rule 8
rule 1 /^9\(1..........\)/ /\1/
rule 2 /^9614\(.......\)/ /\1/
rule 3 /^9\(.......\)/ /\1/
rule 4 /^9\(011.*\)/ /\1/
rule 5 /^9\([2-9]11\)/ /\1/
voice translation-rule 9
rule 1 /^8\(1..........\)/ /\1/
rule 2 /^8614\(.......\)/ /\1/
rule 3 /^8\(.......\)/ /\1/
rule 4 /^8\(011.*\)/ /\1/
rule 5 /^8\([2-9]11\)/ /\1/
voice translation-rule 91
rule 1 /^614\(.*\)/ /9\1/
rule 2 /^\(..........\)/ /91\1/
rule 3 /^\(.*\)/ /9\1/
voice translation-profile SIP
translate calling 7
translate called 8
voice translation-profile cid_fix
translate calling 91
voice translation-profile default
translate called 1
voice translation-profile strip_9_out
translate called 9
voice translation-profile voip
translate called 2
class-map match-any voice-signaling
match dscp af31
class-map match-any voice-udp
description Class Mapping for VoIP RTP
match dscp ef
policy-map voice-qos
description VoIP QoS
class voice-udp
priority 256
class voice-signaling
bandwidth 128
class class-default
interface Dialer1
service-policy output voice-qos
dial-peer voice 800 voip
translation-profile outgoing SIP
destination-pattern 9[2-9]11
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 801 voip
translation-profile outgoing SIP
destination-pattern 9614.......
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 802 voip
translation-profile outgoing SIP
destination-pattern 9[2-9]......
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 803 voip
translation-profile outgoing SIP
destination-pattern 1[2-9].........
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 804 voip
translation-profile outgoing SIP
destination-pattern 9011T
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 799 voip
translation-profile incoming voip
translate-outgoing calling 91
incoming called-number 614xxxxxxx
dial-peer voice 806 voip
translation-profile outgoing SIP
destination-pattern 1614.......
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 807 voip
translation-profile outgoing SIP
destination-pattern 614.......
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 808 voip
translation-profile outgoing SIP
destination-pattern 1[2-9].........
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 809 voip
translation-profile outgoing SIP
destination-pattern 011T
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 810 voip
translation-profile outgoing SIP
destination-pattern *..
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 811 voip
translation-profile outgoing SIP
destination-pattern *1
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 812 voip
translation-profile outgoing SIP
destination-pattern *67.......
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 813 voip
translation-profile outgoing SIP
destination-pattern *671..........
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay sip-notify rtp-nte
codec g711ulaw
dial-peer voice 814 voip
translation-profile outgoing SIP
destination-pattern *123
session protocol sipv2
session target dns:neptune.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw
authentication username 1614xxxxxxx password xxxxxxxxxxxxxxxxxxx
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
mwi-server dns:neptune.vtnoc.net expires 3600 port 5060 transport udp unsolicited
registrar dns:neptune.vtnoc.net expires 3600
sip-server dns:neptune.vtnoc.net
load 7960-7940 P00308000100
max-ephones 48
max-dn 192
ip source-address port 2000
auto assign 1 to 1
service phone displayIdleTimeout 00:30
service phone displayOnDuration 1:00
timeouts interdigit 8
system message CME 4.0
url services »phone-xml.berbee.com/menu.xml
time-zone 12
time-format 24
voicemail *123
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
transfer-system full-consult
transfer-pattern .T
after-hours block pattern 1 1900 7-24
after-hours block pattern 2 1976 7-24
ephone-dn 30 dual-line
number 1614xxxxxxx
label VoIP 614-xxx-xxxx
ephone 1
mac-address XXXX.XXXX.XXXX
paging-dn 31
type 7960
button 1:21 2:29 3:30

FIOS/Viatalk with Actiontec router and PAP2T

Actiontec default settings for it's ports is 100 speed Full Duplex. The PAP2T (and most VoIP equipment) is set to use 10 Half Duplex. If both were set to Auto this would be fine, but they're not. Set the Actiontec to Auto and things should work OK! Or if you know how, set the PAP2T to Full Duplex.

6. Who are some providers?

Some providers of VoIP

  • net2phone:
    net2phone says that with Windows operating systems, you can call any net2phone-enabled PC for free; call any U.S. phone for two cents per minute (U.S. customers); and call world-wide for very reasonable rates. Download free software to participate.

  • Vonage:
    Vonage provides a hardware interface, and unlimited nationwide calling for a current fee of $24.99 monthly for premium, $14.99 basic, plus setup fees. (More on Vonage rates here.)

  • Packet 8
    Packet8 is the name of 8x8's internet protocol telephone service that enables voice and video communications over any broadband connection. The service combines the Packet8 SIP service platform with client technology from Netergy Microelectronics, Inc. and software technology from Centile, Inc. (both 8x8 subsidiaries).
    Packet8 enables enterprise and residential customers to subscribe via a web portal, and delivers advanced IP dialtone services, including IP video, to 8x8 client devices and softphones.

  • CBeyond Communications:
    CBeyond Communications is a next-generation communications company focused on serving the small business market. "We are setting a new protocol by utilizing next-generation Internet Protocol technology to build one of the world's first 100% IP-based communications networks capable of carrying both voice and data services over T-1's. We've partnered with Cisco Systems to build one of the nation's first 100% Internet Protocol networks. This network will provide your business with phone and Internet services in an entirely new, more economical way. With our IP network, we're able to provide both voice and data traffic over the same network."

  • Innomedia:
    innomedia is another provider which enables free device-to-device calling, as well as global low cost calls to any phone in more than 200 countries. Hardware-based devices allow customers to use a telephone and any type of IP connection, from a simple dial-up service to broadband access service delivered via either cable, DSL, fixed wireless, or Local Area Network (LAN).

  • SipTelecom Inc.
    SipTelecom provides a free-of-charge VoIP service. They intend to cover expenses in portal revenues, through business increase resulting from free iPNs and from banner advertising during the call setup time.
    Customers can receive their personal identity similar to their e-mail address, enabling them to receive calls from their web page or an email signature file. They can also create their own phonebook and own answering machine.

  • Free World Dialup
    FWD allows you to make free phone calls over the Internet using a 'regular' telephone or a computer program. The FWD service has over 12,500 subscribers from over 65 countries. You can join today, at no cost, by either using programs already on your computer, or by downloading a free program. Later, an IP phone upgrade will let you realize the total freedom FWD provides whether on Cable, DSL, Dialup modem, or WiFi around town.

  • MultiTech Systems MultiVOIP
    MultiVOIP voice over IP gateways provide toll-free voice and fax communications over the Internet or Intranet. By integrating voice and fax into an existing data network, businesses can realize substantial savings on inter-office long distance toll charges.

  • VoicePulse
    * No computer, software, or configuration necessary!
    * Use your regular touch-tone phone and your existing Internet connection
    * Call anywhere in the United States for free with the America Unlimited and Business Unlimited calling plans
    * Calls to other VoicePulse customers are always free
    * Incoming calls are always free


Here's a pretty informative thread on this provider that should answer most questions.

7. Vonage Information

Why is there so much Vonage information here?

Because, although there are numerous providers of VoIP, Vonage seems to be the clear favorite in the VoIP forum, and that brings many curious people looking for additional information about Vonage in particular.

Where can I get started with Vonage?

Vonage has a speed test page to establish that your high-speed connection will perform satisfactorily.

Click here to test.

Note: If you have a firewall in operation, you will have to permit this test.

How do I set it up?

After you have signed up, Vonage will send you a CISCO ATA-186 which you will need to install.

You then plug any standard telephone into the ATA 186. You'll need a router if you want to be able to use your broadband connection while your telephone is in use.

Note: Many users have reported unacceptable latency problems when using a portable telephone with this system.

And what's really happening when I use it?

Instead of actually talking over a pair of copper wires, as in the standard telephone system, VoIP simply converts your voice from an analog to a digital signal and sends it over your broadband connection as data. It is routed over the internet and to your intended destination where it is treated again as a standard telephone transmission ... your called party sees no difference, if your connection is fast enough.

When you are the called party, your caller dials your number which, because it is a VoIP number now, is routed over the internet and through the Vonage network to your Cisco phone adapter causing your phone to ring.

What features are available?

The Vonage system offers the following features with their service. See this page for more.

Phone Features:

• Call Waiting
• Caller ID
• Voice Mail
• Call Forwarding
• Caller ID Block (*67)
• Repeat Dialing
• Online Account Management
• Real Time Billing information
• Any area code of your choice (See available area codes)
• Lifetime Telephone Number
• Take your Vonage home phone anywhere
• Free Phone Adapter
• Free calls to any other Vonage DigitalVoice Subscriber
• 30-day trial
• International Rates
• Online Support
• Real Time Service Announcements
• Refer-A-Friend program

If the Vonage servers are down, what happens to incoming calls?

If I use Vonage and my broadband connection goes down, will the callers get through to my voicemail and be able to leave a message, or will they receive a busy/dead line signal?

By default callers will be routed to your voicemail when the vonage servers cannot contact your Cisco ATA-186 box. Alternatively, you can set up an secondary number to use when your connection is down (eg: a POTS number). You set this up using a web interface to your Vonage account.

Can I send/receive FAXes with it?


However ... most people report having limited success with faxing. If faxing is an essential service to you, consider an online fax service such as efax instead. Faxing will work over Vonage, but can take a couple of tries.

Where do I sign up?

To sign up with Vonage, go to this page (secure page).

There, you will select from a $39.99 per month unlimited calling plan, or a $25.99/month Unlimited Local/Regional Plus Plan.

There is more information about Vonage rates here.

You may also select an area code of your choice, from the following available area codes:

San Francisco/Silicon Valley/San Jose and surrounding Bay Area
(415) (408) (510) (650) (707) (831) (925)

Los Angeles/Central California
(310) (323) (213) (818) (714) (805)
(661) (562) (626) (949) (909) (760)

San Diego Area
(619) (858)


Miami - Southeastern Florida
(305) (561) (786) (954)

(404) (678) (706)


(815) (847) (773) (708) (630) (312)

Boston Metro
(781) (617) (508) (978)

New Jersey:
North/Central Jersey
(732) (201) (908) (973)

South Jersey

New York:
New York City Metro
(212) (917) (646) (718) (347)

Long Island, Westchester and Upstate
(516) (631) (914) (845)

Philadelphia Area

Pittsburgh Area:

Special email address for DSLR users

Vonage has set up a special email box for users of this site to communicate questions, suggestions and to report any service issues.

The email address is: dslreports@vonage.com

8. Problems

Vonage and Cordless Phones

There have been several reports over time that certain cordless telephones can interfere with the Vonage service and the ATA-186.

If you are having problems and have a cordless, it is suggested that you consider experimenting without the cordless, to see if it is causing the problem.

It is also reported that this can occur with corded phones as well. The real problem is where the base of the phone is too near or on the ATA. Separating the ATA from the phone usually resolves the problem

Can my VoIP service be used with my existing home wiring?

It might be possible but it can present several problems. Altering your home wiring (at your phone jacks) may be necessary, and you may experience problems with some (or all) of your phones not ringing, and possibly risk damaging your phone or VoIP adapter.

For more information see Brian McConnell's excellent article on Telecom Tips - Wiring Your Home For VoIP Service , and this one: How to Distribute VoIP Throughout a Home.

9. Glossary

VoIP Terms

ANI Automatic Number Identification

A telephone function that transmits the billing number of the incoming call (Caller ID, for example).

MGCP Media Gateway Control Protocol

A protocol for IP telephony that enables a caller with a PSTN phone number to locate the destination device and establish a session.

BLI Busy Lamp Indicator

A light or LED on a telephone that shows which line is in use.

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.

DID Direct Inward Dialing

The ability to make a telephone call directly into an internal extension without having to go through the operator.

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).

Diff-Serv Differentiated Services

The Diff-Serv model divides traffic into a small number of classes to provide quality of service (QoS).

PSTN Public Switched Telephone Network

The worldwide voice telephone network.

DNIS Dialed Number Identification Service

A telephone function that sends the dialed telephone number to the answering service.

SIP Session Initiation Protocol

A protocol that provides telephony services similar to H.323, but is less complex and uses less resources.

DTMF Dual-Tone Multifrequency

The type of audio signals generated when you press the buttons on a touch-tone telephone

T1 A 1.544-Mbps

point-to-point dedicated digital circuit provided by the telephone companies consisting of 24 channels.

E1 The European counterpart to T1

which transmits at 2.048 Mbps

TAPI Telephony API

A programming interface that allows Windows client applications to access voice services on a server.


An IEEE standard for providing QoS using three bits (defined in 802.1q) to allow switches to reorder packets based on priority level.

TOS Type of Service

A method of setting precedence for a particular type of traffic for QoS.


An IEEE standard for providing virtual LAN (VLAN) identification and QoS levels. Three bits are used to allow eight priority levels, and 12 bits are used to identify up to 4,096 VLANs.


A communications channel between two points, typically referring to large-bandwidth telephone channels between switching centers that handle many simultaneous voice and data signals.


An ITU standard for real-time interactive voice and videoconferencing over LANs and the Internet

VoIP Voice over (TCP/) IP

A method of sending voice information over a packet-switched network, such as the Internet, using TCP/IP.

Source: Internetwk.com

RTP - Real Time Transport

The Real-time Transport (RTP) Protocol provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level transla tors and mixers.

H323 - Expanded Info

The H.323 standard provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS). These networks dominate todays corporate desktops and include packet-switched TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies. Therefore, the H.323 standards are important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communications. It includes parts of H.225.0 - RAS, Q.931, H.245 RTP/RTCP and audio/video codecs, such as the audio codecs (G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263) that compress and decompress media streams.

Media streams are transported on RTP/RTCP. RTP carries the actual media and RTCP carries status and control information. The signalling is transported reliably over TCP. The following protocols deal with signalling:

RAS manages registration, admission, status.
Q.931 manages call setup and termination.
H.245 negotiates channel usage and capabilities.
H.235 security and authentication.

10. Useful Links

VoIP Gateway Vendors

Voice over IP Gateway Vendors

Protocols.com - Detailed explanations and info

Very detailed and in-depth, but easy to understand explanations of all VoIP aspects.

A Database of IP Voice Systems suppliers

IP Voice Systems This site has been closed.

Primus Canada TalkBroadband FAQ

Contains provider-specific information in addition to a wealth of general VoIP tips, tricks, and troubleshooting.

»Primus TalkBroadband VoIP FAQ