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Voice communications on IP networks use owner protocols as well as standards like SIP, MGCP and H323. In this paper we propose a new method for transparent traversal of NATed (Network Address Translated) networks for the SIP (Session Initiation Protocol) protocol. Although SIP is an application layer protocol, its operation is affected by address translation. This is because SIP uses network layer information (source IP and source port) that is lost by the NAT operation. The suggested method adapts dynamically one of the three solutions: Connection-Oriented media STUN or TURN depending on the situation occurring during call initiation.
Articles accepted for publication will be licensed under the Creative Commons BY-NC-SA. Authors must sign a non-exclusive distribution agreement after article acceptance.
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ISSN
1666-6038 (Online)
1666-6046 (Print)