IP telephony

In IP telephony, Voice over IP or VoIP, the internet or another IP network is used to transport speech. This makes telephony possible on data networks and creates the opportunity to merge the previously traditionally separate worlds of speech and data. This only requires infrastructure once, and new products and services can also be developed. Working with VoIP-based telephone exchanges has become common practice within companies. The term “VoIP” is often misused to refer to the actual transmission of sound rather than its protocol.

IP telephony, Voice over IP or VoIP

VoIP is not the same as VoDSL. With VoDSL, a virtual circuit is configured over an ADSL line without using the Internet protocol TCP / IP.

In IP telephony, Voice over IP or VoIP, the internet or another IP network is used to transport speech. This makes telephony possible on data networks and creates the opportunity to merge the previously traditionally separate worlds of speech and data. This only requires infrastructure once, and new products and services can also be developed. Working with VoIP-based telephone exchanges has become common practice within companies. The term “VoIP” is often misused to refer to the actual transmission of sound rather than its protocol.

VoIP is not the same as VoDSL. With VoDSL, a virtual circuit is configured over an ADSL line without using the Internet protocol TCP / IP.

History

IP telephony has been around since the beginning of the first computer network. The first voting packages were sent by 1973. The technology to do this was available to users from 1980. In 1996, a “Vocaltec Internet Phone Release 4” program came on the market along with other features such as voicemail and caller ID. But it was not perfect yet because the program could not make calls to the older telephone lines but only to the same vocaltec users. This technology was developed in 1998, making calling to everyone possible.

Benefits

An advantage is that an internal company telephone number can be linked to an IP address, so that a teleworker can also be reached at home via his internal telephone number. Previously, this could only be done by forwarding the number to the home address, which entails costs. By means of a Hosted VoIP solution, for example, it is also possible to register a telephone number based on a username and password, not on the basis of an IP address.

Quality

The signal quality may be better than traditional analog telephony if the underlying digital connection is good. In fact, early users sometimes found the quality to be “too good” because the line seemed “dead” due to the lack of noise. To solve this, VoIP phone systems use ‘Comfort Noise’, a system that does nothing more than add a slight noise to the signal to the end user.

The “signal quality” is different from “call quality”. The “call quality” is determined by the method by which the sound is digitized and encoded. A piece of software or hardware that implements this function is called a codec (in this case: speech codec). Some codecs code so efficiently that very little bandwidth is required, but the sound quality (call quality) is therefore less, and comparable to mobile telephony (mobile phone), for example.

Cabling

Office environments can suffice with less cabling because the computer and telephone can be connected to one port. The phone requires a network port to connect the computer, or a separate Ethernet switch must be used. Most VoIP phones have a built-in 1-port Ethernet switch that also transmits voice and data traffic via different VLANs. The speed is often limited. Gigabit Ethernet has barely made its appearance with adapters (ATAs) and IP telephones. The average speed is no more than 100 Mbit / s, with some devices not even going above 10 Mbit / s. The latter are often intended for the home environment, where the speed to the internet has limitations anyway – and that slower speed is no problem. Since 2009, there are a number of brands on the market that offer devices with a gigabit connection (1000 Mbit / s).

Usually the power supply of a VoIP phone is provided by the switch to which it is connected via Power over Ethernet or PoE. This saves an adapter and associated power cable at the workplace. Standard PoE switches can supply at least 15 watts per connection, but there are switches that can supply up to 60 watts for, for example, tablets or video conference phones or powerful WiFi access points.

To make it complete, the switch should be powered by an Uninterruptible Power Supply (UPS) that can continue to provide power even if the regular power supply is down.

Services integration

Integration of different services can become easier with Voice over IP, for example when working together on a document during a conversation or video conference. More important motives will include CTI – think of links with databases – and shared address books and / or telephone books. The additional capabilities and flexibility that VoIP offers may be more important than the potential cost savings that are achievable.

Cost reduction

Companies with multiple locations can make optimal use of fixed connections between locations with IP telephony by using the same bandwidth for voice traffic and data traffic. This provides a cost advantage.

Because existing data connections can be used, the price is usually lower than via a traditional telephone connection. This is interesting for international calls or for calls between different branches of a company.

Accessibility

The Achilles’ heel of traditional telephony is the line between the business center and the local exchange. Everything can be lost if a connection fails. The VoIP application uses standard network protocols that can be treated as normal data traffic. For example, it is possible to connect several data connections to a network, these connections must be realized via separate cableways, so that the connection to the outside world is not broken in the event of a single line failure.

Location independent

In analog telephony, a telephone number is linked to a location. With VoIP, a phone number is linked to a VoIP account (used in hardware or software). The location of the VoIP account is arbitrary. A VoIP phone can therefore function at any location where the phone is connected to the internet.

Devices can also be easily moved within companies, without this affecting the telephone number or the functioning of the device.

Cons

Interference susceptibility

Traditional data networks have traditionally not been designed for the purpose of voice transport. The transport of speech places strict demands on the network with regard to, among other things, the delay that the speech is experiencing and the disruption of the connection. The delay in particular can be a nuisance when it comes to a ‘smoothly flowing conversation’.

With the analog telephone network, a thunderstorm on the way could disrupt the connection. This influence was excluded, among other things, with the use of ISDN and more modern network technologies such as fiber optics. Traditionally, data networks have as their most important property the reliable and relatively fast transport of data from sender to receiver. Some variation in the time it takes to send information is less important. If a packet of data is lost, it is sent again. This contradicts the requirements of a voice network – after all, there is no time to resend it, so packet loss can only be very limited.

With IP telephony, packets of data are sent from the sender to the receiver and back, whereby the delay and reliability are of the utmost importance. The delay that packets experience (Jitter) and their reliability (the amount of packets lost in the network) therefore determine the perception of the call quality of the users. The crowds on the network can negatively impact VoIP traffic. The user may notice this in the call quality. To meet this, mechanisms are added to data networks that enable Quality of Service (QoS). This gives priority to voice traffic over data traffic.

The IP protocol used on the Internet has been given a mechanism in the design which makes it possible to distinguish between the various types of information that are transported over a connection. The ability to apply QoS has been embedded in the protocol itself for years. This option can be used by an internet connection provider, but it is not (yet) commonplace. For example, links (peering) between networks of providers often do not yet use this option, also because handling data traffic in this way requires extra powerful (and expensive) equipment.

Current ADSL and cable modems and routers increasingly have this QoS function built-in as standard, so that when connecting to the Internet you can influence the handling of traffic, so that for example the download of a file on the PC does not have the quality of affects the telephone conversation. The exact implementation differs from manufacturer to manufacturer, the goal is always the same: to get the best VoIP traffic over the connection.

Power supply

Telephones that use Voice over IP often need their own power supply (adapter). Traditional analog devices do not need these, because the telephone signal and the power are supplied over the telephone line. This makes Voice over IP devices less suitable for emergency lines. Most VoIP hardware is now suitable for Power over Ethernet (PoE). The network cable is used as a power supply. The network equipment should also support PoE in such a case, which affects the price.

Location independent

Additional measures must also be taken with regard to location independent if the telephone exchange is not in the same physical location as the device. For example, when calling the area-code-free national emergency number, the emergency center in the country and region of the telephone must be called, not the emergency center in the region of the telephone exchange. Modern IP telephone exchanges offer the option of sending an alternative sender number (caller ID) or region code when calling the emergency number.

Application

Consumers

VoIP is used in two variants for consumers, which are generally referred to as digital telephony and internet telephony. In contrast to digital telephony, internet telephony uses the internet. Digital telephony and internet telephony are expected to largely replace the existing PSTN connections of consumers for telephony.

Digital telephony

Digital telephony is a term that is generally used when talking about VoIP via your own internet provider. Around 2000, ISDN had that name. In 2006, the existing internet providers expanded their product range with fixed telephony subscriptions. The existing ADSL or cable modem is hereby replaced by a copy with a VoIP connection. A traditional telephone can be connected to the modem. The voice traffic to the telephone exchange remains within the (controlled) network of its own internet provider, so that it can be given priority (if any) over other data traffic. This makes it easier to control the call quality and accessibility. Usually only regular telephone numbers can be called with digital telephony subscriptions. For this, this provider has a link (peering) with the public telephone network.

Internet telephony for fixed and mobile

Internet telephony stands for telephony via the public Internet. It is also possible to take out a telephone subscription with a provider other than your own internet provider. Voice traffic then only does not remain within the network of its own internet provider, but goes over the internet to reach the telephone exchange. As a result, no quality guarantee can be given. In some cases, the same ADSL modem can be used as supplied by the Internet service provider. This is not possible in the situation with cable modems. Then additional hardware is used (for example a router with VoIP connection, ATA, IP telephone) or a software telephone on the computer in combination with a sound card, headphones and microphone. Well-known VoIP providers with software phones (softphones) are Windows Live Messenger, Google Hangouts, Skype, Viber and VoipBuster. Well-known softphone programs are X-Lite and SJphone, which simply offer a software SIP telephone that can in principle work with the service of any VoIP provider.

VoIP can also be used on smartphones, depending on the type and the telecom provider, in the form of VoIP via the mobile phone.

If you want to be able to reach any telephone, not just one running a special program that allows you to establish a direct peer-to-peer connection, you can have a VoIP provider connect the Internet to the Internet for a fee. telephone to be called. The compensation often depends only on the country of destination and whether it concerns calls to a fixed or mobile number. The country from which you call does not matter.

“Internet calling”

Special internet calling telephone numbers that are actually a kind of e-mail address work for internet telephony and not for digital telephony. An example of such an internet calling telephone number is: “3112345@domai.co.uk” or “johnchapman@domain.co.uk”. The exchange of such a provider is then not linked to the normal telephone network and there is no need to be related to the country / city codes as used in the public telephone network. For example, this could be a system that works under Asterisk. An example of a service that is offered is Free World Dialup (no longer free since August 2008), an initiative to make IP telephony available in order to test its usability in practice. Originally, it was only possible to make calls between subscribers to that service. There are now also links with other providers, via peering, which extends the possibility to make calls using only the internet costs. Calling these numbers is completely free, because the telephone traffic is completely transported via the internet and the provider does not have to invoice because he is only responsible for the transport of the data packets. There is still a telephone exchange as an intermediary, which takes care of making / breaking the connection. Completely without compensation for a VoIP provider, calling VoIP is direct from IP address to IP address, without the intervention of a third party. A conversation conducted in this way therefore requires no. The disadvantage of this is that IP addresses are not static in all cases, which means that someone is suddenly no longer available when the address changes. There must also be some technical facilities available for this to work. Most Internet telephony equipment that uses the SIP standard has an IP-IP calling feature.

Companies and telcos

Internet telephony is already widely used in business. The telephone exchanges of various (foreign) branches are connected to each other via the internet. Calls between the branches no longer need to be made via the PSTN network. The employee does not notice this.

Telephone companies in the wholesale carrier market use VoIP a lot for routing international calls. Only the beginning and end of the connection is via the traditional telephone network.

For smaller companies or partnerships with employees working in different locations or from home, Hosted VoIP solutions where the (company) telephone exchange is purchased as an online service are often an efficient solution.

Radio amateurs

Radio amateurs know VoIP in the form of the Echolink and D-Stars system, in which VoIP is combined with transceivers, so that you can wirelessly ‘step on’ the Internet VoIP system.

Technic

VoIP requires special telephones or exchanges that can be connected to the internet. There are also manufacturers who supply modems that can convert VoIP to ISDN equipment via a so-called S0 connector.

The SIP protocol is a common VoIP protocol used to set up a connection. Cable operators usually use a special protocol, the “packet cable protocol”, which is integrated into the cable modem, which is also used for cable internet. This protocol ensures that speech takes precedence over data.

The following obstacles have been overcome to properly communicate VoIP:

  • Because the network does not offer certainty that the packets are delivered in the correct order and whether or not they are delivered, VoIP experiences problems with delays and thunder. Especially if it is routed via a satellite, this can be up to 600 milliseconds. The receiver must then reconstruct packets that arrive late, out of order or not at all, to ensure that the sound remains smooth. This is often done by building a bounce buffer into the device (home gateway, ATA or IP telephone) or into the network equipment. Recent versions of Asterisk have options for activating the jitter buffers.
  • The next obstacle in successfully sending packets is the firewalls and addresses. Solutions have also been found for this. On the one hand, session controllers can be used that run alongside the firewall and enable VoIP calls in highly secured networks. On the other hand, Skype has designed the proprietary protocol to route calls through Skype peers on the network and bypass firewalls. Other methods include using other protocols, such as STUN or ICE.
  • Quality of service (QoS) is a technique for prioritizing voice data packets on the network over less priority data packets.
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