Introduction to Design and Implementation of Sip for Multistreaming Applications Project:
The Session Initiation Protocol (SIP) is for the signaling protocol used for the preparation and differentiation of the Multimedia type of communications like voice calls and video calls in the field of Internet. The possible applications are like video conferencing, streaming of multimedia distributions, online games, emails etc. The SIP has applications like creation, modification, termination of the two way party or Unicast and Multiparty or Multicast areas of more than one media fields.
The Proposed topic constitutes of developing the applications for the text and file transfer, voice and video conference by the use of SIP along with proxy server system. The SIP signaling is based on the pattern of Server- Client which uses protocols HTTP or SMTP. The requirement of the project includes the Internet phone or soft phone. The Internet phone applies the Session Initiation Protocol or Media Gateway Protocol (MeGaCo).
The SIP phone consists of the VoIP phone. The Soft Phone runs on the personal computer and does not need any special device. The soft Phone enhances the quality and similar to the normal phone along with head set or can be used with the USB device. The applications of SIP phone includes Voice conference, IP contact, IP phone system, FAX on the IP, Video conference, Call monitoring.
The modules is needed to develop are
- User Agent Client or UAC which develops the conclusion and transfer to the servers.
- User Agent Server or UAS which receives the requests and process that requests to create response.
The system has to be developed in the Linux 2.6 kernel operating system, ANSIC, GCC compiler, GDB debugging tool. The hardware requirements are 2 personal computers, 512 MB RAM, CPU with 2.2 GHz or above, LAN connection.
Download Design and Implementation of Sip for Multistreaming Applications Project Report .