A new methodology to adapt SIP Protocol for voice traffic transported over IP Network

摘要

The convergence of company communications on IP network continues to oscillate between the protocol owners and the standards SIP, MGCP and H.323. We propose in this paper a new approach allowing a transparent traversal of NATs (Network Address Translation) to SIP protocol (Session Initiation Protocol). This ensures thus optimisation in the case of multimedia sessions. Indeed, the fact that SIP belongs to the application layer constitutes a weakness vis-àvis the traversal of NATs. It is due, on the one hand, to the way in which the server responds to requests of clients. On the other hand, it is caused by the dynamic allocation of the UDP ports. The approach proposed, called ''Adequate Solution for each Situation'' (ASS), allows to adapt in a dynamic way one of the following three solutions: connection-oriented media, STUN and TURN, following the situation which occurs during the call initialisation.