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A Prototype of a VoIP/PSTN Gateway on the basis of RFC 3398
In the recent several years, there has been a surge of interest in IP telephony, especially actors such as Skype and Google Talk have earned much publicity in media. However, in spite of this interest for IP telephony or Voice over IP (VoIP), traditional Public Switched Telephone Networks (PSTN) will most likely play an important role in several years to come, thus it vital that the two networks are able to seamlessly interwork.
Interworking between VoIP and PSTN entails on one hand that the media, i.e., the voice traffic, is able to flow between the two networks, something that requires a conversion between the A-law format of PSTN to G.711 and the Real-time Transfer Protocol (RTP); on the other hand, it entails that the adhering signaling traffic is able to traverse the two networks, which, among other things, requires a conversion between Integrated Services digital network User Part (ISUP) call signaling and Session Initiation Protocol (SIP) signaling. To address the latter, the Internet Engineering Task Force (IETF) has proposed a mapping between ISUP and SIP signaling in RFCs 3398 and 3578.
Your thesis work includes designing a prototype VoIP/PSTN gateway which accommodates both media and signaling conversion. The gateway has to support A-law (PSTN) to G.711 (IP, RTP) media conversion, and should implement signaling conversion in accordance with IETF recommendation RFC 3398.
1.Make a preliminary design of a gateway prototype and document the design in a design specification.
2.Implement, in an iterative way, RFC 3398. Each iteration should encompass at most two call scenarios from RFC 3398.
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