Metaverse: a SIP/SIMPLE IP telephony and instant messaging compliant system

ABSTRACT

The growing utilization of Internet Broadband access is popularizing Instant Messaging services enhanced with Voice and Video. However, despite the existence of open and standardized signaling protocols, the most popular services still use proprietary protocols and clients, hindering the interoperability among the existent solutions. This paper presents the methodology of implementation and validation of Metaverse: an IP Telephony and Instant Messaging application compliant with the Internet Engineering Task Force (IETF) open specifications, allowing its interoperability with other standards adherent solutions.A crescente utilização de Internet Banda Larga vem popularizando os serviços de Mensagens Instantâneas com recursos de conversação via voz e vídeo sobre IP. Entretanto, apesar da existência de protocolos de sinalização abertos e padronizados, os serviços mais populares utilizam protocolos e clientes propriet´rios, impedindo a interoperabilidade entre as diversas soluções. Este artigo apresenta a metodologia de implementação e validação do Metaverse: uma aplicação de Telefonia IP e Mensagens Instantâneas totalmente aderente às especificações abertas do Internet Engineering Task Force (IETF), permitindo sua interoperabilidade com outras soluções aderentes aos padrões.

REFERENCES

• 1: D. Cohen, Specifications for the Network Voice Protocol (NVP), RFC Editor, 1977
• 2: Lennox, J. e Schulzrinne, H., Feature Integration in Internet Telephony, Proceedings Feature Integration in Telecommunications and Software Systems VI, Glasgow, United Kingdom, Maio de 2000. URL: http://www.cs.columbia.edu/~hgs/papers/Lenn0005_Feature. pdf. Acesso em 10/2005.
• 3: Packetizer, A Primer on the H.323 Series Standard". URL: http://www.packetizer.com/voip/h323/papers/primer/. Acesso em 11/2005.
• 4: J. Rosenberg , H. Schulzrinne , G. Camarillo , A. Johnston , J. Peterson , R. Sparks , M. Handley , E. Schooler, SIP: Session Initiation Protocol, RFC Editor, 2002
• 5: Baset, Salman A. e Schulzrinne, H. "An Analysis of the Skype Peer-to Peer Internet Telephony Protocol". Columbia University, 2004. URL: http://www1.cs.columbia.edu/~salman/skype/. Acesso em 11/2005.
• 6: JABBER. "Extensible Messaging and Presence Protocol (XMPP)". Request for Comments 3920, Internet Engineering Task Force, 2004. URL: http://www.ietf.org/rfc/rfc3920.txt. Acesso em 11/2005
• 7: NIST. About the IP telephony project. URL: http://snad.ncsl.nist.gov/proj/iptel/. Acesso em 07/2006.
• 8: IVOV, Emil. "A Jain-SIP Video Phone for the People!" http://snad.ncsl.nist.gov/proj/iptel/nist-sip-communicator.html. Acesso em 07/2006.
• 9: Janjanak. "K-Phone". URL: http://sourceforge.net/projects/kphone. Acesso em 07/2006.
• 10: Yamamori M. e Esaki H. Approach of VoIP/SIP Interoperability Task Force. Journal of the National Institute and Communications Technology Vol 52, 2005. URL: http://www.nict.go.jp/publication/shuppan/kihou-journal/journal-vol52no3.4/04-04.pdf. Acesso em 07/2006.
• 11: B. Campbell , J. Rosenberg , H. Schulzrinne , C. Huitema , D. Gurle, Session Initiation Protocol (SIP) Extension for Instant Messaging, RFC Editor, 2002
• 12: Rosemberg J. "A Presence Event Package for the Session Initiation Protocol (SIP)". Request for Comments 3856, Internet Engineering Task Force, 2004. URL: http://www.ietf.org/rfc/rfc3856.txt. Acesso em 11/2005.
• 13: Sugano H., Fujimoto S. e outros. "Presence Information Data Format (PIDF)". Request for Comments 3863, Internet Engineering Task Force, 2004. URL: http://www.ietf.org/rfc/rfc3863.txt. Acesso em 11/2005.
• 14: Harvey M. Deitel , Paul J. Deitel, Java How to Program, Prentice Hall PTR, Upper Saddle River, NJ, 2001
• 15: Ranganathan, Mudumbai e O'DOHERTY, Phelim. JAIN SIP Tutorial. Sun Microsystems, 2003. URL: http://dns.antd.nist.gov/proj/iptel/tutorial/JAIN-SIP-Tutorialv2.pdf. Acesso em 05/2006.
• 16: Rivieres, J. e Beaton, W. Eclipse Platform Technical Overview. IBM, 2006. URL: http://www.eclipse.org/articles/Whitepaper-Platform-3.1/eclipse-platform-whitepaper.html.
• 17: Dhulipalla, S., Meyer, G. e ROOS, C. SIP Interoperability Scenarios Test Plan (Draft). International Multimedia Teleconferencing Consortium, 2000. URL: http://www.imtc.org/interops/test_plans/sip_interop_draft.do c. Acesso em 05/2006.
• 18: Schulzrinne, H. Classification for SIP Interoperability Test Event. SIPit, 2000. URL: http://www.cs.columbia.edu/sip/sipit/classification.html. Acesso em 05/2006.
• 19: Brookshire. Asterisk - The Open Source PBX! Digium, 2005. URL: http://www.asterisk.org/about. Acesso em 05/2006.
• 20: Kuthan, J., Janak, j. e outros. SIP Express Router - Admin Guide. FhG Fokus, 2002. URL: http://www.iptel.org/ser/doc/seruser/seruser.html. Acesso em 04/2006.
• 21: J. Rosenberg , J. Weinberger , C. Huitema , R. Mahy, STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs), RFC Editor, 2003

INDEX TERMS

General Terms:
Algorithms, Experimentation

Keywords:
SIP, VoIP, mensagens instantâneas, normatização, simple, telefonia IP