Voice over IP – also known as VoIP – is a technology used to transmit voice over a network intended for data transmission. Put simply, this is Internet telephony, because with VoIP you can make calls using special IP telephony devices or traditional telephones with adapters via computer networks, ie the Internet . The advantages are obvious: VoIP is a cost-effective alternative to the traditional telephone connection. The only requirement for VoIP telephony is a stable and fast DSL connection, so you get a good telephone connection via the Internet.

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What is  VoIP / IP telephony/  Internet telephony?

IP telephony is the common voice communication over the Internet Protocol and is also referred to as Voice over IP (VoIP) or Internet telephony. Using IP telephony, data and telephone calls can now be routed over one and the same network. In contrast to analogue telephone connections, IP connections are linked to broadband , thus allowing simplified telephony over the Internet connection. However, for this it is first necessary to find a suitable VoIP provider. What to look for when choosing your future VoIP provider, you can find out here .

As the new technology offers network operators and users many advantages , conventional analogue and ISDN telephony is losing more and more importance and will ultimately be completely converted to VoIP telephony.

To make calls via VoIP over the Internet connection, you have various options. The purchase of additional hardware is not required in most cases. For example, you can use your PC, laptop or smartphone to make a phone call. For more information,

Differences Between Voip And Traditional Telephony

 Voip vsTraditional Telephony

Basically, the functionality of VoIP does not differ from that of traditional telephony: It takes place first the connection, then the call transfer and then the disconnection instead. In contrast to traditional telephony, no separate line is switched through for the telephone call, but the call is transmitted in digitized form in data packets via the Internet protocol. For this purpose, the speech signal of the telephone operator is detected at a specific sampling rate and the results are converted by means of an analog-to-digital converter (ADC) into a regular sequence of digital signals.

The data rate can be reduced by coding, with many compression techniques relying on omitting irrelevant information for the human ear. The amount of data is thereby reduced without noticeably impairing the quality of the audio impression. However, if compressed too much, the voice quality also suffers.

Due to transmission disruptions , dropouts or echoes, voice quality does not match that of traditional telephone networks, but the quality is generally better than that of mobile network connections. Who has a good DSL connectionhas now received an equivalent and cheaper alternative to the traditional telephone line. If the VoIP technology is coupled with a conventional telephone system, incoming calls still require a telephone number. For outgoing calls, however, no phone number is necessary. Here, the internal telephone number (SIP address) is displayed.

Voip Transmission Quality Depends On The Provider

The communication quality of a VoIP call requires that the data packets used arrive at the telephone receiver so that they can be reassembled according to the original speech signal. The claimed providers are crucial to the transmission quality of the transmission. How much data can be transmitted in a certain period of time (the so-called throughput) depends on the coding used. The data rate of an uncompressed call is usually 64 Kbps.

For pure IP telephony, a maximum bandwidth of approximately 100 Kbit / s is required. The 64 Kbit / s overheads of the different communication protocols are calculated. Overheads are data that the shipping itself requires. Because the Internet connection is principally used for various applications, the connection of a private user should have at least a bandwidth of 100 kbit / s in both directions. In general, however, the upstream bit rate is significantly lower than that of the downstream. With connections over VoIP it can come also to the package loss, whereby a loss rate of up to five per cent is still regarded as acceptable.

How does Voip / IP Telephony work?

how does voip work

In IP telephony, the phones are no longer plugged into the TAE socket of the telephone jack, into an ISDN box or into a splitter, but into the DSL router or into a VoIP-enabled cable modem . Also in the background, a completely different technical process takes place in the Internet telephony. The connections are no longer assigned fixed lines – instead, like any other digital service, the language is sent in the form of data packets over the Internet. This is usually prioritized due to the real-time requirements. Also a telephone call, which is led over the InterNet, can be divided like in the classical Telefonie into three basic procedures:

(1) call setup, (2) call transfer, and (3) call termination

a. How does the connection establishment and dismantling work with VoIP?

In contrast to voice transmission, another protocol is used for IP-based telephony for setting up and breaking down VoIP connections, namely the Session Initiation Protocol (SIP). This network protocol ensures manufacturer-independent integration of VoIP components. For SIP-based systems, each participant has their own SIP address.

Example of a SIP addressThis is similar to building an e-mail address. For example, a SIP address might look like this: “sip: 0123456744@example.com”.
A SIP address consists of two components. On the one hand it contains the SIP user of the participant and on the other hand the domain name of the registrar server.

In order to establish a connection, it is necessary for the caller (A-subscriber) to know the IP address of the recipient. For this, the terminals of the participants in the conversation register with their IP address, their user name and their password at a registrar server . Thus, the user is the location-independent calls possible, because he can log on to any SIP terminal worldwide, access all his phone services and keep his phone number. Prerequisite for this is only an Internet connection.

If now a connection between two participants come about, the terminal of the subscriber A sends a message with the phone number of the subscriber B to the server of his provider A. Then the server gives the information to the server of the provider B on, so that the terminal of the participant B can appeal. If this step works without any problems, terminal B rings and sends a message back to terminal A. This indicates that the other party has been found and answers with a beep. When establishing a successful connection, the communication then takes place between the terminals and no longer via the SIP server.

To terminate the call transfer, a terminal sends a corresponding SIP packet to its server. The server then notifies the other terminal of the disconnection and the call is terminated.

b. How does the call transfer work?

As already mentioned, VoIP telephony no longer transmits analog voice information but individual digital data packets . For this to be possible, the acoustic signals, which are recorded analogously by a microphone, must first be transformed into analog electrical signals. These are then digitized and divided into many small packages. Thereafter, the data packets are sent via a public or a private network.

For the transmission of data, the type of coding is crucial, because this determines the quality of the conversation. Some codecs, in contrast to the standard G.711, serve to achieve the lowest possible data rate. Codecs like G.722 are chosen for HD-quality phone calls . Although they ensure excellent voice quality, they also require a much higher data rate.

Once the data has arrived at the recipient’s terminal, they must be decrypted in a last step, because only then, as humans, is it possible for us to understand the information received. Phones and softphones usually automatically choose the best quality codec on both sides.

What are the requirements for VoIP or IP telephony?

requirements for IP telephony

To use the Internet telephony, you only need three things:

  1. An internet connection
  2. A VoIP provider
  3. Suitable hardware

Since you are making calls over the Internet with an IP connection, an Internet connection (eg DSL) with sufficient bandwidth is essential to ensure excellent voice quality. Approximately 100kbit / s of bandwidth will accumulate in the upload and download direction per voice channel.

In addition to an Internet connection, some hardware is also required to use VoIP. However, it should be said straight away that it is not necessary to acquire additional hardware. There are 4 ways to make calls via VoIP, which we would like to introduce below:

1. VoIP by PC

The computer or laptop is connected to the Internet via a broadband connection. In order to be able to make calls via the PC, it is necessary to install suitable software, a so-called softphone . Softphones are available in different versions and also as free variants. We at Placetel highly recommend the free VoIP softphone Zoiper . In addition, the computer must have a microphone and a speaker. However, it is recommended to use a headset for making telephone calls when there is a high volume of traffic.

2. VoIP via IP phone

Another way to make calls via VoIP, is the use of an IP phone (desk phone, wireless phone or DECT phone ), for example, the manufacturer Snom , Yealink or Cisco . IP Phones are similar in appearance to traditional phones, but differ in the technique used. IP phones support data transmission over the Internet. You connect the IP-based telephone to the Internet via a free LAN port of your router (for example, FRITZ! Box from AVM).

3. VoIP via analog phone

If you do not want to buy new SIP phones or make calls via the PC , you can simply keep your analogue or ISDN telephones used so far. However, for this it is necessary to purchase an analogue telephone adapter (ATA) or an ISDN telephone adapter (ITA) . The adapter offers on the one hand the possibility to connect the phone and creates on the other side a connection possibility to a DSL modem and thus to the Internet. The adapter then converts the analog signals into digital data packets and vice versa. Other options we have listed in our blog article ISDN telephone system to VOIP connection .

4. VoIP over the smartphone

Picture of a man holding a smartphone in the hand.
The last option is to use your smartphone for Internet telephony.

You can simply download an app for your iPhone or Android Phone and save on the purchase of an additional IP phone. Learn more about Mobile VoIP here .

Benefits For VoIP Communication

In addition to the conventional transmission of telephone calls, telephone network operators in Europe can also transmit calls using IP telephony. In this case, only parts of the network or the entire network can be used for this purpose. The cost advantages resulting from the worldwide transmission of telephone calls are used and passed on by call-by-call providers for the provision of international calls.

Security Of Voip Data Transmission

Since VoIP data packets are sent over a network, which in turn is used by several subscribers, it can not be ruled out that attackers can pick up these packets and record the call. Computer programs can be used to record data streams and generate audio files. An encryption of the VoIP is difficult, because sometimes the voice quality is disturbed. In IP telephony, the SIP (Session Initiation Protocol) is often used, which serves to negotiate the communication modalities between two participants.

Despite its security barriers may be attacks on this protocol be successful – for example through denial-of-service attacks. VoIP telephony is also used for trick fraud on the Internet such as the so-called Vishing (Voice Phishing). In the case of automated telephone calls, the victims are asked to publish sensitive data such as bank accounts or passwords. Anyone who is prompted by phone call, should end the call and call back to the respective company.

Voip  Alternatives To Skype And Whatsapp

One of the best known forms of VoIP is Internet telephony via Skype. In addition to WhatsApp and Facebook Messenger, there are also a variety of other software, with which you can also make free calls over the Internet. These include, for example, ooVoo, Windows Live Messenger, Google Talk, Team Speak and Mumble.

With ooVoo video chats and the writing of text messages with up to twelve contacts are possible simultaneously and it can parallel video messages and files are sent to multiple contacts. In addition, you can easily integrate chats into websites and blogs. One feature that ooVoo Skype has in mind is the ability to connect to Facebook, which can be used to create a video chat to your own Facebook contacts. ooVoo is also available on Macbooks,

After Microsoft has replaced Windows Live Messenger with Skype, you can access your Skype contacts directly from your Windows Live account (Hotmail). Video telephony is also possible directly from the e-mail account.

This is also the case with Google Talk, but just in relation to an e-mail account at the Internet giant Google. Skype alternatives that are particularly suitable for gamers are Team Speak and Mumble. These consume very few resources, so that parallel running programs are not disturbed by the Internet telephony. Nevertheless, a good voice quality is ensured during these calls while talking on the phone.