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What's VOIP
Voice over Internet Protocol (VoIP) is a protocol optimized
for the transmission of voice through the Internet or other
packet switched networks. VoIP is often used abstractly to
refer to the actual transmission of voice (rather than the
protocol implementing it). VoIP is also known as IP
Telephony, Internet telephony, Broadband telephony,
Broadband Phone and Voice over Broadband. "VoIP" is
pronounced voyp.
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.
Pre-Paid Phone Cards
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.
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