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In telephony "Voice over IP" (Internet telephony) identifies a group of technologies for the transmission and delivery of vocal calls through a data network connected to the Internet or a local area network. More specifically the VoIP acronym implies for the signal to be packed then transmitted in the form of IP packets, the usual mean of transmission for digital data, instead of being transmitted directly as for traditional telephony.
VoIP telephony is an alternative to traditional one which can be integrated recurring to dedicated devices. In the past two decades a steady number of companies started offering VoIP related products and services. Thanks to these companies it is now possible to make and receive calls to / from a traditional telephone network and to use cheaply priced advanced features like conference calls and video calls.
Analog to Digital Conversion
The first step to use an IP network to transmit a voice signal consists of voice digital encoding since Internet protocols work with digital data only. A traditional analog to digital converter such as the ones used in standard ISDN telephony for the last couple of decades can take care of the encoding. For VoIP it is not necessarily a hardware device; the conversion can be performed by a computer software running on a standard PC.
The resulting digital stream of data can be split and arranged in packets transmitted in a standard and secure way along other data streams like files, music, images, multimedia or programs.
Conversely to what happens with ISDN technology, where the signal is immediately transmitted through a line, VoIP introduces an additional step to filter and further encode data in order to remove noise and suppress useless information leaving only the meaningful one. For example pauses are removed. Statistics reveal that in a conversation between two parties each of the participants remain silent 50% of the time. With traditional telephony the line is busy "transmitting" silence; filter programs instead remove the unnecessary information enabling substantial bandwidth savings. Background noise is removed too making it easier to understand your partner even when you are surrounded by confusion.
Finally data are compressed recurring to audio codecs in order to attain further bandwidth savings. Compression too is executed after digital encoding because it's impossible to compress an analog sound signal.
When filter operations are concluded data are arranged in packets and finally transmitted recurring to the better fitted Internet protocol available. The UDP protocol is usually used because telephony does not work well with delays or high latency caused by retransmitting lost packages.
Client / Server Architecture
Two entities interact in traditional telephony:
- Telephones or fax machines.
- A telephone company or a local PBX.
In VoIP those entities are replaced by:
- Clients, which can be telephones or other devices like a PC with a microphone and head-set.
- A server controlling phone calls the same way a local PBX or a telephone company does.
The client duties include signal encoding / decoding and packet transmission while the server performs traffic control functions for clients, call signaling, device identification, device management, etc. .
We already talked about data transfer. Packets are sent and received through a low level IP protocol: usually UDP. Signalling and control functions are managed through protocols too, sadly more than a single standard protocol exist. A lot of protocols can be used both proprietary, developed by hardware producers to meet their needs, and open ones, developed and maintained by independent committees. We'll ignore proprietary protocols because configuring devices of different producers to talk to each other is a very difficult task.
Several open protocols exist:
- H.323 backed by the International Telecommunications Union
- - It's the older of the bunch and its use has been in decline for years.
- - Originally developed for videoconferencing, the protocol was adapted to VoIP telephony, but some issues remain deriving from its roots.
- - Its affected by the same intercommunication problems of proprietary protocols.
- SIP (Session Initiation Protocol) backed by the Internet Engineering Task Force
- - Developed for real-time Internet telephony.
- - It's the de facto standard for VoIP telephony.
- - Guarantees maximum interoperability between devices.
- IAX2 (Inter Asterisk Xchange 2) developed by Digium for Asterisk
- - Used by the Asterisk Open Source PBX.
- - Developed for inter server communication.
- - Has good performances for client / server communication as well as for server / server.
- - Open source. Supported by a main developer and a great number of third party developers and hardware producers.
VoIP adoption promotes a convergence of voice and data traffic. A voice dedicated network becomes redundant and unnecessary, replaced by a local network. An entirely VoIP based system requires the presence of a network infrastructure and broadband connectivity to work well. The network can be built out of any combination of networking technologies:
- Fiber channel.
With but a single constraint to keep in mind: performances. In other words the sum of planned data traffic and concurrent voice calls or video calls the system should support once in production. A similar reasoning should concern routers, switches and all other apparatuses making the network. Selection constraints consist of performances and the speed needed to support the wanted number of connected users and their day to day activities.
The number of planned concurrent calls constraint should be kept in mind when sizing the link to the outside world too. An analog 56 Kb or ISDN link could be enough to support a single call or a couple were ISDN in use, but a larger number of concurrent telephone calls requires for a faster link. You'll need a DSL or Fiber channel link.
The present paper provided an introduction the the internals of a VoIP telephone network. For further information about the technology, its strong points and the issues related to a migration You are kindly invited to contact us using the following contacts.
To contact me or leave me your feedback, Please e-mail at studiosg [at] giustetti [dot] net.
Languages: English - Italiano