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Abstract:-
Generally we use telephone network in order to have transfer of voice(media) but the
telephone ID are getting exhausted and need for the circuit switching in telephone network
makes it less efficient and results in utilization of large amount of resources, hence forth we
attempt to transfer voice/media using packet switching with the help of IP(internet
protocol) and PBX(private bank exchanging). An audio communicating session among three
or more people who are geographically dispersed is provided by a conference function in a
PBX or multiline telephone or by the telephone companies. See voice chat,
videoconferencing and data conferencing. Basically a private branch exchange is a
telephone exchange that serves a particular business or office, as opposed to one that a
common carrier or telephone company operates for many businesses or for the general
public. The Internet Protocol (IP) is the principal communications protocol used for relaying
datagrams (packets) across an internetwork using the Internet Protocol Suite. Responsible
for routing packets across network boundaries, it is the primary protocol that establishes
the Internet.
We also attempt to make a three people conferencing using voice over internet protocol
(VOIP), it mainly involves signaling and conferencing. Voice over Internet Protocol (VoIP) is a
general term for a family of transmission technologies for delivery of voice communications
over IP networks such as the Internet or other packet-switched networks.
IP PBX is an application of VOIP technology. it is used inside a campus to provide voice and
video communication. Separate protocols are used for call signaling and transport of
voice/video in packet form. audio conferencing is one important feature provided by an IP
PBX.
Project topic
This project implements 3-party audio conferencing. The conference connection is assumed
to be established and project deals with mixing of voice.
Voice samples from each conference participant is sent to a conference server (mixer) using
RTP protocol. RTP protocol uses UDP and IP protocols. Mixer uses a suitable logic to choose
one of the three RTP streams and sends it back to all three .participant whose stream is
selected will ignore received stream, to avoid listening to his/her own voice. For other two,
received stream will be played. we will use pre-recorded voice samples stored in .wav file.
Scope of project
The project involves design, coding, and testing of software modules to implement
conferencing feature. C- Language is used for coding.
Tools used
GNU gcc compiler, Id link, and gdb source level debugger.
PSTN-Public Switch telephone network switch used circuit switching in order to transfer
media/voice calls.
Voice mail system-Used to store data in case the subscriber wishes to do so.
Data networking infrastructures-Involves the seven layers IP used to transmit the data.
The benefits of audio conferencing are numerous. First and foremost, audio conferencing
can lower transportation costs. Commuters spent $60 billion dollars on gas driving to
business meetings and other work-related appointments in 2003.
Secondly, with audio conferencing, a meeting can be set up in only a few minutes. In the
real world, planning can sometimes take months!
Audio conferencing is a whole lot cheaper than other means of conferencing. It can reach
more people since it relies mostly on telephone lines. It is easier to use since it takes
advantage of a technologythat is relatively familiar to all of us. Furthermore, we can enhance
it by combining communication lines with other mediums such as computers. Audio
conferencing has a great potential for larger groups to conference in and the technology used
is fairly reliable. Audio conferencing is also very accessible and consumes lesser time to set
up.
Security:Audio conference access is protected through the use of conference IDs, unique
participant PINs and passwords.
Project by,
Vishwanath.M (1AM07EC080)