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What's VOIP ?
Companies providing VoIP service
are commonly referred to as providers, and protocols which are used to
carry voice signals over the IP network are commonly referred to as Voice
over IP or VoIP protocols. They may be viewed as commercial realizations
of the experimental Network Voice Protocol (1973) invented for the ARPANET
providers. Some cost savings are due to utilizing a single network to
carry voice and data, especially where users have existing underutilized
network capacity that can carry VoIP at no additional cost. VoIP to VoIP
phone calls are sometimes free, while VoIP to public switched telephone
networks, PSTN, may have a cost that is borne by the VoIP user.
Voice
over IP protocols carry telephony signals as digital audio, typically
reduced in data rate using speech data compression techniques,
encapsulated in a data packet stream over IP.
There are two types of
PSTN to VoIP services: Direct Inward Dialing (DID) and access numbers. DID
will connect the caller directly to the VoIP user while access numbers
require the caller to input the extension number of the VoIP user.
History
Voice over Internet Protocol has been a subject
of interest almost since the first computer network. By 1973, voice was
being transmitted over the early Internet.[1] The technology for
transmitting voice conversations over the internet has been available to
end users since at least the 1990's. In 1996, a shrink-wrapped software
product called Vocaltec Internet Phone Release 4 provided VoIP, along with
extra features such as voice mail and caller id. However, it did not offer
a gateway to the analog POTS, so it was only possible to speak to other
Vocaltec Internet Phone users.[2] In 1997, Level 3 began development of
its first softswitch (a term they invented in 1998); softswitches were
designed to replace a traditional hardware switchboards by serving as the
gateway between two telephone networks.
Functionality
VoIP can facilitate tasks and provide
services that may be more difficult to implement or expensive using the
more traditional PSTN. Examples include:
* The ability to transmit more
than one telephone call down the same broadband-connected telephone line.
This can make VoIP a simple way to add an extra telephone line to a home
or office.
* 3-way calling, call forwarding, automatic redial, and
caller ID; features that traditional telecommunication companies (telcos)
normally charge extra for.
* Secure calls using standardized protocols
(such as Secure Real-time Transport Protocol.) Most of the difficulties of
creating a secure phone over traditional phone lines, like digitizing and
digital transmission are already in place with VoIP. It is only necessary
to encrypt and authenticate the existing data stream.
* Location
independence. Only an internet connection is needed to get a connection to
a VoIP provider. For instance, call center agents using VoIP phones can
work from anywhere with a sufficiently fast and stable Internet
connection.
* Integration with other services available over the
Internet, including video conversation, message or data file exchange in
parallel with the conversation, audio conferencing, managing address
books, and passing information about whether others (e.g. friends or
colleagues) are available online to interested parties.
Security
Many consumer VoIP solutions do not support
encryption yet, although having a secure phone is much easier to implement
with VoIP than traditional phone lines. As a result, it is relatively easy
to eavesdrop on VoIP calls and even change their content.[9] There are
several open source solutions that facilitate sniffing of VoIP
conversations. A modicum of security is afforded due to patented audio
codecs that are not easily available for open source applications, however
such security through obscurity has not proven effective in the long run
in other fields. Some vendors also use compression to make eavesdropping
more difficult. However, real security requires encryption and
cryptographic authentication which are not widely available at a consumer
level. The existing secure standard SRTP and the new ZRTP protocol is
available on Analog Telephone Adapters(ATAs) as well as various
softphones. It is possible to use IPsec to secure P2P VoIP by using
opportunistic encryption. Skype does not use SRTP, but uses encryption
which is transparent to the Skype provider.
The Voice VPN solution
provides secure voice for enterprise VoIP networks by applying IPSec
encryption to the digitized voice stream.
VoIP has become an important
technology for phone services to travelers, migrant workers and
expatriates, who either, due to not having a fixed or mobile phone or high
overseas roaming charges, choose instead to use VoIP services to make
their phone calls. Pre-paid phone cards can be used either from a normal
phone or from Internet cafes that have phone services. Developing
countries and areas with high tourist or immigrant communities generally
have a higher uptake.
Technical details
The two major competing standards for
VoIP are the ITU standard H.323 and the IETF standard SIP. Initially H.323
was the most popular protocol, though in the "local loop" it has since
been surpassed by SIP. This was primarily due to the latter's better
traversal of NAT and firewalls, although recent changes introduced for
H.323 have removed this advantage.
However, in backbone voice networks
where everything is under the control of the network operator or telco,
H.323 is the protocol of choice. Many of the largest carriers use H.323 in
their core backbones[citation needed], and the vast majority of callers
have little or no idea that their POTS calls are being carried over
VoIP.
Where VoIP travels through multiple providers' softswitches the
concepts of Full Media Proxy and Signalling Proxy are important. In H.323,
the data is made up of 3 streams of data: 1) H.225.0 Call Signaling; 2)
H.245; 3) Media. So if you are in London, your provider is in Australia,
and you wish to call America, then in full proxy mode all three streams
will go half way around the world and the delay (up to 500-600 ms) and
packet loss will be high. However in signaling proxy mode where only the
signaling flows through the provider the delay will be reduced to a more
user friendly 120-150 ms.
One of the key issues with all traditional
VoIP protocols is the wasted bandwidth used for packet headers. Typically,
to send a G.723.1 5.6 kbit/s compressed audio path requires 18 kbit/s of
bandwidth based on standard sampling rates. The difference between the 5.6
kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth
optimization techniques used, such as silence suppression and header
compression. This can typically save 35% on bandwidth usage.
VoIP
trunking techniques such as TDMoIP can reduce bandwidth overhead even
further by multiplexing multiple conversations that are heading to the
same destination and wrapping them up inside the same packets. Because the
packet header overhead is shared between many simultaneous streams, TDMoIP
can offer near toll quality audio with a per-stream packet header overhead
of only about 1 kbit/s.
Aujourd’hui après des
années de tâtonnements, la téléphonie sur Internet et la VoIP sont sorties des
cercles d’amateurs de nouvelles technologies pour toucher à la fois les
entreprises et les résidentiels. La téléphonie à prix discount sur Internet en
est encore à ses débuts sur les marchés des entreprises et résidentiels.
Téléphonie : Téléphoner moins
cher (SIP - H323 - PABX - IPBX)
Le protocole SIP brigue la téléphonie Internet :
Lancé avec une offensive marketing qui l’a plutôt desservi, le protocole SIP n’a
pas percé dans la téléphonie IP, face à H.323. Il constitue pourtant un socle
ouvert et standardisé, porteur de nouveaux services réseaux, couplant
informatique et téléphonie, en entreprise comme dans le monde des opérateurs.
Depuis dix ans l’industrie des communications a été témoin de profonds
changements dans le besoin de communication entre l’individu et les
organisations ou les entreprises. Beaucoup de ces changements ont été le fruit
de l’explosion d’Internet et de solutions applicatives basées sur le même
protocole de transport IP. Graduellement ce furent les communications e-mail
externes puis les communications e-mail internes, suivies par tous les éditeurs
ou développeurs d’applications qui franchirent le pas IP. Ces derniers y
voyaient un moyen unique et simple de s’affranchir de barrières jusqu’alors
propres à des mondes propriétaires (Téléphoner gratuitement serait désormais
possible).
Plus récemment l’Internet s’est étendu partiellement dans l’Intranet de chaque
organisation, voyant le trafic total basé sur un transport
réseau de paquets (IP) surpasser le trafic traditionnel du réseau voix
(réseau à commutation de circuits) (DataQuest,1998).
Il devenait clair que dans le sillage de cette avancée technologique, les
opérateurs, entreprises ou organisations et fournisseurs devaient pour
bénéficier de l’avantage du transport unique IP, introduire de nouveaux services
voix et vidéo. Ce fût en 1996 la naissance de la première version
voix sur IP appelée H323.
Issu de l'organisation de standardisation européenne ITU-T sur la base de la
signalisation voix RNIS (Q931), ce standard a maintenant donné suite à de
nombreuses évolutions, quelques nouveaux standards prenant d’autres orientations
technologiques.
*
Comprendre les tendances de l'informatique.