In today's environment nearly all end-to-end telephone connections are set up via circuit-switching using Public Switched Telephone Network (PSTN), whereby node-to-node links in an origin/destination connection are set up via interconnects,and the connection is maintained exclusively for exchanges of information between the origin and destination until it is torn down.An alternate way of setting up end-to-end connections that is widely used for transmission of data is packet-switching,whereby origin-to-destination connections are effected by node-to-node,store-and-forward relay of small segments of data sets that are reassembled at the destination;this technique is called Voice over Internet Protocol (VoIP).VoIP is considered as the third generation of telecommunication telephony after the analog and digital telecommunication technology. This thesis study the architecture of packet-switched telephone networks and then analyzes the structure of VoIP technology,which is the Transmission Control Protocol/Internet Protocol (TCP/IP) model,some protocols reside in the application layer (i.e.,Session Initiation Protocol (SIP)for call control,Session Description Protocol (SDP)for description media stream,and Real-time Transport Protocol (RTP)for media exchange). Over years,Interest of security is increasing.To provide privacy for user's conversation in VoIP,there is a need to implement a security for media transmission.A Secure Real-time Transport Protocol (SRTP)is designated to provide security for real-time media transmission using an encryption method, but it does not provide key agreement between participants.This thesis implemented SIP protocol and a key agreement using pre-shared key protocol within SDP protocol used by SIP.The implementation is done using UNICON language.