Frequently Asked Questions

VoIP services convert your voice into a digital signal that travels over the Internet. If you are calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless "hot spots" in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable you to use VoIP service wirelessly.

In most cases, yes.

Some VoIP service providers offer services that can be used wherever a high speed Internet connection available. Using a VoIP service from a new location may impact your ability to connect directly to emergency services through 9-1-1.

Only if your service requires you to make calls using your computer. All VoIP services require your broadband Internet connection to be active.

If you have a special VoIP phone or a regular telephone connected to a VoIP adapter, the phone will ring like a traditional telephone. If your VoIP service requires you to make calls using your computer, the software supplied by your service provider will alert you when you have an incoming call.

In June 2005 the FCC imposed 911 obligations on providers of “interconnected” VoIP services – VoIP services that allow users generally to make calls to and receive calls from the regular telephone network. You should know, however, that 911 calls using VoIP are handled differently than 911 calls using your regular telephone service.

Problem:

How fast does my connection to the Internet need to be in order to use the service?

Solution:

Your connection speed (bandwidth) must be sufficiently high for the type of vocal codec you are using. Vocal codecs (or just "codec") are the standard means by which your IP Phone (UA) encodes your voice and transmits it across the Internet. The codec that is picked for a particular phone call is primarily determined by your UA's configured preferences. Consult your UA's manual to determine how to change codec preferences.

To use the best quality / highest bandwidth codecs (G711a/u) we recommend a connection speed of no less than 90Kb/s in each direction (up and down).

Problem:

What digits do I need to dial to make an International phone call?

Solution:

To dial any PSTN phone number that is outside of the NANP (North American Numbering Plan), you must dial:

011 + Country Code + City Code + Phone number

Problem:

Sometimes the voice of the person I am speaking to sounds choppy or broken up. Why? What can I do?

Solution:

This can be caused by a variety of problems. The most common is poor Internet connection quality or lack of bandwidth.

Very often what is sold as a "High Speed" or "Broadband" connection is not up to the task of transmitting voice packets in a timely or reliable manner. Problems such as high latentcy (slow delivery of packets) or packet loss (actual dropping of voice data) can be seen from time to time on some provider's networks. These problems will be heard by you on your IP phone call long before they are noticed for other data activities like checking email or web browsing.

To track down where the problem is coming from, you can use packet tracing software to reveal network bottlenecks and get them reported to your ISP. We recommend a product called "Ping Plotter". You can download a free trial version at http://www.pingplotter.com

Problem:

Can I send a fax from my account to a traditional fax machine(IP to PSTN)?

Solution:

IP to PSTN faxing is supported.

Problem:

Will VoIP work with a Satellite Internet connection?

Solution:

In general, we do not recommend the use of our system over a Satellite (SAT) Internet connection. However, some customers do. There are a few technical issues related to SAT links that can cause problems:

- Round Trip Time (RTT) Delay. Most SAT's used for Internet connections are geosynchronous. This means that their orbit period is the same as the length of a day, so they are stationary in relationship to a given point on the ground. To achieve such an orbit, the SAT must be 22,235 miles (35,784 km) above the earth's surface. Radio waves travel at the speed of light, but even the speed of light takes a noticeable amount of time to travel that distance 4 times (up and back twice for a complete round trip from your UA to our servers.) This delay, when added to other factors on the ground, can cause delays anywhere from 500 ms to 1000 ms (1/2 second to 1 second.) It can be very difficult to carry on a conversation when there is that much delay.

- Network Jitter. Probably more important than RTT Delay itself is the consistency (or lack of) of the RTT Delay. Some packets take longer than others to reach their destination which can cause voice data to arrive out of order and possibly be dropped. We have found that many SAT Internet connections have this failing. Many UA's have adaptive Jitter Buffers which can absorb some small inconsistencies in RTT delay, but given large inconsistencies, the result is often an almost unusable connection.

- Lack of uplink bandwidth. Typically SAT Internet connections are asymmetrical with their download bandwidth being much higher than their upload bandwidth. If your uplink bandwidth is lower than the bandwidth required by the vocal codec you are using (see the Knowledge Base article on Codecs), then you will not be able to use the connection for voice calls reliably.

Problem:

Sometimes when I try to place a PSTN call, I get a recording stating:

"An error has occurred. Please contact Technical Support with error code 104."

Solution:

The problem is that the system has detected more than one IP Phone (UA) registered with a phone number that is associated with an unlimited calling plan. Calling plans are restricted for use by only a single UA to prevent abuse. The simple solution is to make sure that you only have one UA programmed with the virtual phone number that is associated with your calling plan.

Occasionally, due to power outages or UA reboots more than one registration can appear for the same UA, triggering an error 104. The old registration will eventually expire (you can check for when on your control panel.)

Problem:

Sometimes when I try to place a PSTN call, I get a recording stating:

"An error has occurred. Please contact Technical Support with error code 205."

Solution:

The system has detected a PSTN call from an IP Phone (UA) that is not registered correctly. To prevent abuse, all UA's must register with our proxy server before they can place a PSTN call. Common reasons for your UA not to be registered are:

PROBLEM: Incorrectly set USERID (phone number) or PASSWORD in the UA

SOLUTION: Make sure your USERID and PASSWORD are set correctly. Re-enter them if necessary.

PROBLEM: "Registration" is set to "Never" or "No" in the UA's configuration

SOLUTION: Modify your UA's configuration to force it to register before placing a call.

PROBLEM: Call placed too soon after registration

SOLUTION: Wait at least 5 seconds after successful registration before making a PSTN call.

Problem:

Can I send a fax from one user phone number to another (IP to IP)?

Solution:

This FAX service is not guaranteed.

IP to IP faxing, both the sender and receiver need to be CIP Telecom members or the other providers to support fax and have their fax machines attached to IP analog telephone adapters (ATAs).

One other important factor when faxing over VoIP is the quality and speed of your Internet connection. Since fax data cannot be compressed, the G711u/a CODEC must be used. This codec requires a minimum of 64Kb/s in both directions to be reliable, but we recommend more than 90Kb/s.

Problem:

What is the registration interval and what should I set it to?

Solution:

Registration is the event where your IP Phone (UA) contacts our system and tells it where you can be reached for incoming calls. The registration interval is the time (measured in minutes or seconds) between when your UA re-registers. Registration is required before you can make PSTN calls (see the Knowledge Base article on Error 205.)

Our sip proxy forces a 60 minute registration regardless of the interval specified in your VoIP adapter.

Problem:

What does 'PSTN' mean?

Solution:

PSTN is an acronym for Public Switched Telephone Network. It is the traditional telephone system that the world has been using for decades to make phone calls. In order to access the PSTN from your IP Phone (UA), you need to use a "bridge" between VoIP and PSTN.

If you are a Premium Member, this is accomplished seamlessly by dialing a 1 + area code + phone number for a North American call or 011 + country code + city code + phone number for an International call.