a general depiction of realizing voice communication through the Internet.
The basic VoIP components are made up by specific protocols informing both
parties as well as any intermediary (gateway, VoIP provider and so on) on
the kind of content to be transmitted (either audio or video), how the call
to be both commenced and terminated, along with the digitalization as well
as compression-decompression codec to be used.
Up until a while ago, when incoming-call providers facilitating VoIP applications
started making their appearance, communication between VoIP users and any fixed
telephone communication system had not been feasible.
Communication between two VoIP applications or appliances,
initially involves support by common protocols and codecs at the same time.
In case such a communication is rendered unlikely to be accomplished, it is
possible that an in-between provider will offer a codec conversion of some
sort, so that communication is carried out [Transcoding].
Real-Time Transport Protocol [RTP] & Real-Time Control Protocol
is practically used in any audio-video transmission by all networks. It has
taken on the-as far as possible-correct audio-video packets transmission
from point A to point B, avoiding thus delay mishaps, jitter and other inter-network
awkward situations. It does not carry out any other procedures, like, for
instance, service quality control or network resource management.
SIP & H.323
No more nor less than the two
biggest VoIP players. They both set up a career around 1995, when the need
for audio-video network transmission
showed up. They have undertaken the task of notifying a receiver that there
has been a requirement for communication by the caller. The next step is to
sort out the type of communication, who is going to take part in it and so
on and so forth.
The basic differentiation between the two protocols is focused on the fact
that H.323 is dependent on a relatively “crabbed code”, while the
Session Initiation Protocol [SIP] is based on text instructions, like the HTTP
protocol. The former (H.323) comes from traditional telephone networks and
likewise cooperation with such networks becomes easier, yet the location as
well as handling of any problems-should they arise-is more difficult when compared
to the SIP.
The H.323 is in fact an “umbrella” of protocols specializing in
a range of communication sections [H.225, H.245, T.38 used in fax over IP and
so on), while both are using several different Internet protocols to be able
to cover any gaps coming up when achievement of voice communication is in progress.
Here, for the sake of information, it would be appropriate if we referred to
the Session Description Protocol [SDP], the Resource Reservation Setup Protocol
[RSVP] to ensure high voice quality and of course the UDP/TCP/IP ones, used
to break up packets and route them to the Internet.
Even though there is a sort of a conflict at a development and management providers
level, it is more than evident that the SIP tends to become the model of today’s
Internet voice communication, not to mention the inclusion of video while calling,
when circumstances become favorable.
Other VoIP protocols
Apart from the H.232 and the SIP there are other voice communication protocols
achieved by net infrastructures, like the H.248 and the Media Gateway Control
Protocol [MGCP], which are specialized in gateway communication. Gateways
are the links between a network-ie, the IP-with either a fixed [PSTN] or
telephony network. There are other VoIP protocols too, which are either closed
ones or they cooperate with specific applications-telephone exchanges. For
example, the particularly popular Skype is a restrictive VoIP protocol by
itself, whereas the Inter-Asterisk Exchange [IAX] is being utilized by the
open software telephone exchange. Finally, the Skinny Call Protocol has been
developed by Cisco.