Design and Implementation of SIP Based Embedded VoIP Phone

Abstract

Voice over Internet Protocol (VoIP) is an emerging technology that enables voice communication over the Internet using the Internet Protocol (IP). By integrating multiple modes of communication using this single infrastructure, it is possible to provide a variety of new services. The emergence of the Session Initiation Protocol (SIP) with SDP (Session Description Protocol), RTP (Real Time Transport Protocol) and UDP it is possible to develop PSTN features in IP network. SIP promises simple and efficient handling of multimedia sessions among multiple users.

Objective of Project VoIP at NirmaLabs was to create campus wide VoIP Architecture. In this VoIP architecture main component to be developed was Embedded VoIP Phone. SIP based VoIP software architecture was designed and implemented for Embedded VoIP Phone. SIP UA (User Agent), using SIP, Media and General Library, was implemented. This SIP-based VoIP system provides a wide range of services such as multiple audio encoding formats, and complex functionalities such as call muting and call-hold facilities, and personal mobility support. Also designed and implemented DAC (Digital to Analog Converter) and ADC (Analog to Digital Converter) modules. To make SIP UA compliant with target platform TS-7250, DAC and ADC modules are integrated in SIP UA. Miscellaneous problems generated during different phases of Project VoIP were solved.