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 Technical
details
There is a lot of debate about the two most popular types of VoIP; SIP
and H.323, each of them has its own merits. 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 500-600ms) 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
5.6kbps compressed audio path will require 18kbps of bandwidth based
on standard sampling rates. The difference between the 5.6kbps and
18kbps 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 6-7kbps data
stream.
Protocols
Most standards-based solutions use either the H.323 or Session
Initiation Protocol (SIP) protocols. A number of proprietary designs
also exist.
Signaling protocols:
Session Initiation Protocol (SIP)
defined by the IETF, newer than H.323
H.323
defined by the ITU-T
Megaco (a.k.a. H.248) and MGCP
both media gateway control protocols
Skinny Client Control Protocol
proprietary protocol from Cisco
MiNET
proprietary protocol from Mitel
CorNet-IP
proprietary protocol from Siemens
IAX
the Inter-Asterisk eXchange protocol used by the Asterisk open source
PBX server and associated client software
Skype
a proprietary peer-to-peer protocol used in the Skype application
Jajah
a proprietary peer-to-peer protocol used in the Jajah SIP and IAX
compatible webphone
Several different speech codecs can be used for stream audio
compression. Commonly used codecs for VoIP traffic include G.711,
G.723.1 and G.729, all ITU-T-specified. |
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