Today, VoIP communication is commonly used to make calls online. The use of VoIP has become so popular because it allows users to make long distance and international telephone calls over their internet connection at a fraction of the cost associated with PSTN and mobile network providers.
With virtual telephone numbers now available from websites, such as mydivert.com, the VoIP telephone can now be used to make and receive calls just as you would with any other traditional telephone. In order to do this the VoIP user can use software on their PC, which will work in a similar manner to skype, with a headset and microphone, or they can use a hardware device such as a VoIP telephone or ATA adaptor.
For VoIP to replace the traditional landline telephone, clear 2-way audio, reliable transmission of DTMF tones, and connection availability are important issues.
The most significant factor that affects the call quality and functioning of the VoIP client is the speed of the internet connection used for making and receiving phone calls. The overall quality and reliability of your VOIP communications are completely reliant upon the quality, reliability and speed of the Internet connection that it uses. The internet speed (or bandwidth) in both UP and DOWN directions is important. A minimum bandwidth of 128 kpbs is recommended for VoIP calls.
Another consideration is the means used to make and receive calls. In general a VoIP hardware device has several advantages, and produces far better results than a software softphone or dialer. Some of the advantages of the hardware based solution include:
# A VoIP hardware device remains available and connected when your PC is turned off.
# A hardware VoIP telephone is not impaired when your PC is under heavy load (running multiple applications, or games etc.)
# Audio for your hardware device is not interrupted by other applications and PC sounds (email arriving etc.)
# SIP Registration is more reliable with a hardware based device.
# Hardware VoIP telephones generally have far more configuration options available.
There are many brands, and types, of VoIP telephone available. These range from single line VoIP telephones to fully functioning pbx systems. Another good solution is the ATA telephone adaptor. These small adaptors, about the size of a cigarette packet, allow you to use a standard telephone for VoIP calls. If you are planning to use just one or two VoIP lines then a simple ATA adaptor will suffice. More information about ATA devices can be found by doing a web search for 'PAP2 ATA'.
In order to make the most from a VoIP line you will need a Direct Inward Dialing (DID) telephone number. This will enable callers to reach you on your VoIP telephone by dialing a standard telephone number. DID for VoIP use are still not available for every country, and all area codes, in the world although the coverage is increasing every day. DID numbers for most of the major business cities of the world are available, and an internet search will soon through up a list of providers.
Direct Inward Dialing (DID) virtual numbers and VoIP work very well together when everything is configured correctly. As most DID/VoIP calls rely on more than one server or carrier working together discrepancies in configurations can result in unstable communication. Common problems include, one way audio, loss of DTMF tones, echo distortion, and dropped calls. These problems generally arise from incompatible codec translations, or other restrictions affecting one or more of the services being used.
If you are planning to use a VoIP carrier it is important to check that an incoming call from your DID is going to be received. You may need to determine the originating IP address for your incoming DID calls and check that calls from this IP are accepted by your carrier. If you are using an Asterisk server, or some other pbx, you may need to add this IP address to the configuration (as an IP authenticated peer).
Check that your DID provider is supplying free, or very low cost, incoming call minutes. A DID with unlimited free incoming minutes, or at least 5000 minutes per month, would be considered reasonable in todays market. Do your research and check the per minute charges if you do exceed the monthly limit.
Check that your DID, VoIP carrier, and PBX are using compatible codecs.
The G711 codec is the most basic, and widely used, telecommunication audio codec. The G711 codec commonly ensures the most reliable transmission of DTMF tones (required for 'ring tones' IVR answering systems), but does not use audio compression (causing distortion where internet bandwidth is restricted).
The G729 codec is also very widely used and produces excellent results for both DTMF and audio quality. However, with the G729 codec a licence is required by the PBX administrator if any transcoding is required.
It has been known for service providers to have an inadequate licence for the G729, restricted to a fixed number of simultaneous transcoded calls. In the circumstance users could experience loss of audio at times of high server usage. If all parties involved in the call route have G729 available then no transcoding is required and the calls will pass straight through.
The use of codecs will be negotiated between the servers involved in relaying your incoming call. The exact codec used will be determined by the codecs available on each server, and the preferred codec order list. Sometimes it may be necessary to experiment with calls to find the optimum configuration.
MyDivert.com - VoIP DID Virtual Numbers
