While communicating with my new customers in last 10 years, The top questions that I faced, were related to Voip Provider . The customer with previous experience with voip face no issues selecting and configuring a Voip Provider however It was really a difficult task for new customer to find out a suitable Voip Provider for his needs . A customer with no background in Voip industry hardly grasp the basics of Voip and consequently he faces difficulty to select a proper Voip Provider.

Finding a proper Voip Provider involves number of technicalities like protocol selection, codec selection, number of concurrent channels , call generation rate, quality issues and compatibility with application. Below we will discuss these issues and will guide new comers to find out a proper voip provider for their needs.

Scope of this article does not allow me to indulge in Voip Basics therefore I am providing related link that will help reader to know more about Basics of Voip. http://www.voip-info.org/wiki/view/What+is+VOIP

Who Are VoIP Providers:

We usually known VoIP provider as VoIP service provider which offer VoIP Internet telephony solutions to residential and commercial customers. They also provide the services and hardware to subscribers under monthly rates. The mechanism of call transmission of VoIP provider is different from traditional Public Switched Telephone Network (PSTN). VoIP provider utilize packet-switched telephony to transmit calls over the Internet it is different from the circuit switched telephony. 

Types of Voip Providers

Voip Providers can be divided into following three main categories

1. Residential Voip

Residential Voip is basically resembles to traditional home phone service where a per-configured IP phone or adapter connected with internet to provide home phone services whereas traditional home phone services connected with PSTN Lines .

2. Business Voip

A Business Voip can be hosted PBX , On Premises PBX or any other kind of Voip Services offered to commercial users.

3. Wholesale Voip

A Wholesale Voip focused to offer services to services providers or customers that interested to deal in bulk Voip like wholesale DID buying , wholesale sip termination or sip origination services.

Key Factors Before Choosing VoIP Service Provider.

As our  focus  is  wholesale  Voip  provider therefore  we will consider important  points  that  you  should  keep in mind  while  finding   and selecting  a wholesale voip provider for your needs. 
1. Capacity requirements.

First you need to estimate the maximum number of concurrent calls that your system can process at a time and after finding out the number of concurrent calls, you need to confirm and demand from voip provider that whether he will fulfill our capacity requirements. It is also called concurrent channels capacity or number of ports required .

As an example , if your have setup a system with 100 concurrent channels / ports then said setup will be capable to dial out 100 outbound calls at a time.

2. Protocol supported

You also need to take care of compatibility issues and need to find out whether technology behind your Voip Provider and Your system is compatible with each other means if your have SIP compliant setup then your Voip Provider also have SIP compliant setup.

In telecommunications, a communication protocol is a system of rules that allow two or more entities of a communication system to communicate between them to transmit information via any kind of variation of a physical quantity.

Following are main protocols used in VoIP

SIP, Session initial protocol

SIP: – Sip is most widely used protocol in Voip industry . SIP offered by IETF and now it is defacto standard of Voip world. SIP treats signaling and data differently also routes them with different routs.

IAX/IAX2 – Inter Asterisk Exchange Protocol was offered by digium and it uses both signalling and data link on same trunk

H323: H323 is ITU Voip protocol.

3. Codecs supported

Codec selection is also play important also both Voip Provider and your setup needs to support same codec to communicate with each other that is why you have to confirm provider about required codecs .

Codec are used to convert our analog voice signal into digital data from transmission from source to destination through internet and these digital data signal reaches destination, these are again converted back to analog signal using codecs and we hear the voice message.

G.711 ulaw (supported in US) , High quality, Low processing but high bandwidth consumption as bandwidth consumption will be 64 kbps per channel .

G.711 alaw ( supported in Europe) ,High quality, Low processing but high bandwidth consumption as bandwidth consumption will be 64 kbps per channel.

GSM GSM is basically a cellular standard codec available free, average quality and low bandwidth requirments 13 kbps

G.729 needs licensing to be use in production. High quality, High processing requirements but lower bandwidth needed about 8 kpbs per channel.

4. CPS

CPS stands for call per second, it is basically call generation rate and you have to verify from provider that your voip account does support required CPS rates. As an example , a 100 concurrent calls setup may have 10 cps.

It is also required to match calls per scond rate of Voip Provider connection with our system, As an example if we have 10 cps account from a voip provider then we should also have full capaicty

5. Origination or Termination

If We are calling other numbers from our system , we will required a Voip Porivder Account that offer Termination services however if others are calling towards our setup then we need origniation services to receive incoming calls.

6. A-Z range

A-Z means alphabetically all countries supported , A Voip Provider with A-Z termination means all destination across the globe are supported and we can call all countries .

7. Competitive call rates

Also you have to compare call rates of Voip Provieder for your required destination, Voip provider will differ with pricing and you have select best option keeping in mind both quality and price.

8. Quality of Service (QoS) Latency / Jitter and Packet Loss

QoS (Quality of Service) is a major issue in VOIP implementations and we have to take all measures to to guarantee that packet traffic for a voice will not be delayed or dropped due interference from other lower priority traffic.

Things to consider are

Latency: Delay for packet delivery, time that voice packets takes from source to destination . ITU-T G.114 recommends a maximum of a 150 ms one-way latency
Following link give detail insight about network delay


Jitter: Variations in delay of packet delivery is called Jitter . Voice packets are transmitted from source sequentially however some packets take more time to reach destination due to different routes they travel to reach destination. The maximum tolerable value of jitter is 50 milliseconds as higher delay will badly effect quality of service.

Packet Loss: Too much traffic in the network causes the network to drop packets, The default G.729 codec requires packet loss should be less than 1 percent

Fore more detail, please visit following links

10. Fax support

If there is need to add support Fax over IP in system then you have too verify from provider that whether it support Fax either G.711 pass through or T.38 . T.38 fax support is more reliable and it is recommended when compared to G.711 pass through .

Here you will find detail comparison of both Fax through G.711 pass through and T.38 faxing.


11. Internet Bandwidth

A G.711 Voip call takes 87 kbps data rate and we have to keep in mind while setting up a Voip based system.

List Of Some Voip Provider Companies those offer wholesale Voip and Foip


Voip Innovations

Flow Route

Alcazar Networks

Omega Telecom

ICTBroadcast blog

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