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Voice over Internet Protocol (also called VoIP
[pronounced "voyp"], IP Telephony, Internet
telephony, and Broadband Phone) is the
routing of
voice conversations over the
Internet or through any other
IP-based network.
Protocols 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.Voice over IP traffic can be deployed on any IP
network, including ones lacking a connection to the rest of
the Internet, for instance on a
local area network.
Technical details:
The two major competing protocols for VoIP are SIP and
H.323. Initially H.323 was the most popular protocol, though
its popularity has decreased in the "local loop" due to its
poor traversal of NAT and firewalls. For this reason as
domestic VoIP services have been developed, SIP has been far
more widely adopted. 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, and
the vast majority of callers have little or no idea that
their POTS calls are being terminated over VoIP. So really
SIP is a useful tool for the "local loop" and H.323 is like
the "fiber backbone". With the most recent changes
introduced for H.323, however, it is now possible for H.323
devices to easily and consistently traverse NAT and firewall
devices, opening up the possibility that H.323 may again be
looked upon more favorably in cases where such devices
encumbered its use previously.
Where VoIP travels through multiple providers' Soft
Switches the concept 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 Signalling 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 signalling proxy mode where only the
signalling flows through the provider the delay will be
reduced to a more user friendly 120-150 ms. These proxy
concepts could lead the way to true global providers.
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 will
require 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
optimisation techniques used such as silence suppression and
header compression this can typically save 35% on bandwidth
used. But the really interesting technology comes from VoIP
off shoots such as TDMoIP which take advantage of the
concept of bundling conversations that are heading to the
same destination and wrapping them up inside the same
packets. These can offer near toll quality audio in a 6-7
kbit/s data stream. |
