Technical details
The two major competing standards
for VoIP are the IETF standard SIP and the ITU standard H.323.
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, 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.
See
also
-ISDN BRI and ISDN PRI Services
-FXO vs FXS
-Global
System for Mobile Communications
-About VoIP
-SIP:Session Initiation Protocol
-List of commercial voice over IP
network providers
-Mobile VoIP
-List of SIP software