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Micro-controller based Internet phone
Author(s)
Kaplan, Shaun
Date Issued
2004
Type
Thesis
Publisher
Cape Technikon
Abstract
This work describes research towards the development of a micro-controller based, standalone
Internet telephone to be used as an alternative to conventional line telephones. Our
definition of 'stand-alone' refers to the unit's capability to perform its function wholly
without the need for an attached computer. The unit should be low cost and capable of
allowing two users to communicate using the units. Bandwidth usage should be kept low
to allow the unit to be used over dial up connections which are prevalent in South Africa.
The units should be easy to use as the anticipated users may be unskilled.
A module containing a 16-bit micro-controller, an Ethernet controller, flash memory
and RAM was chosen as the controller. The module came with a real-time operating
system and a TCPlIP stack.
The session initiation protocol (SIP) was selected to perform the signalling. SIP uses
the session description protocol (SDP) to negotiate the attributes of the media session to
be established.
The real-time transport protocol (RTP) was implemented to transport encoded audio
between the end points. The RTP control protocol (RTCP) was implemented to provide
basic quality of service parameters.
The ITU-T recommendation G.729 annex A was the voice codec selected. Codec ICs
were used to encode and decode the audio.
The implementations were designed specifically for a two user, direct communication
environment. That is two phone units were developed that communicated directly with
each other and not through intermediary servers.
Internet telephone to be used as an alternative to conventional line telephones. Our
definition of 'stand-alone' refers to the unit's capability to perform its function wholly
without the need for an attached computer. The unit should be low cost and capable of
allowing two users to communicate using the units. Bandwidth usage should be kept low
to allow the unit to be used over dial up connections which are prevalent in South Africa.
The units should be easy to use as the anticipated users may be unskilled.
A module containing a 16-bit micro-controller, an Ethernet controller, flash memory
and RAM was chosen as the controller. The module came with a real-time operating
system and a TCPlIP stack.
The session initiation protocol (SIP) was selected to perform the signalling. SIP uses
the session description protocol (SDP) to negotiate the attributes of the media session to
be established.
The real-time transport protocol (RTP) was implemented to transport encoded audio
between the end points. The RTP control protocol (RTCP) was implemented to provide
basic quality of service parameters.
The ITU-T recommendation G.729 annex A was the voice codec selected. Codec ICs
were used to encode and decode the audio.
The implementations were designed specifically for a two user, direct communication
environment. That is two phone units were developed that communicated directly with
each other and not through intermediary servers.
Additional information
Thesis (MTech (Electrical Engineering))--Cape Technikon, Cape Town, 2004
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