SIP links

출처 : http://dpnm.postech.ac.kr/mcs/

References

Papers & Documents

Papers

1. Kundan Singh and Henning Schulzrinne, "Peer-to-Peer Internet Telephony using SIP"
2. David A. Bryan and Bruce B. Lowekamp, "SOSIMPLE: A SIP/SIMPLE Based P2P VoIP and IM System"
3. Salman A.Baset and Henning Schulzrinne, "An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol", Department of Computer Science, Columbia University, New York NY 10027, September 15, 2004
4. Dennis Bergstrom, "An analysis of Skype VoIP application for use in a corporate environment", CISSP, October 2004
5. Xiaotao Wu and Henning Schulzrinne, "Feature Interactions in Internet Telephony End Systems", Department of Computer Science, Columbia University, January 24, 2004
6. Henning Schulzrinne, Sankaran Narayanan, Jonathan Lennox and Michael Doyle, "SIPStone - Benchmarking SIP Server Performance", Columbia University, Ubiquity, March 10, 2002
7. Kundan Singh and Henning Schulzrinne, "Failover and Load Sharing in SIP Telephony", Department of Computer Science, Columbia University
8. Kundan Singh and Henning Schulzrinne, "Unified Messaging using SIP and RTSP", Deptment of Computer Science, Columbia University, NY, USA, October 11, 2000
9. Xiaotang Zhang and Henning Schulzrinne, "Voice over TCP and UDP", Department of Computer Science, Columbia University, September 28, 2004
10. Xiaota Wu, Henning Schulzrinne, "Service Learning in Internet Telephony", Department of Computer Science, Columbia University
11. Joao Paulo Sousa, "An Iptel Architecture Based on the SIP Protocol", Instituto Politecnico de Braganoa
12. Feng Liu, Wu Chou, Li Li and Jenny Li, "WSIP - Web Service SIP Endpoint for Converged Multimedia/Multimodal Communication over IP", Avaya Labs Research, 233 Mt. Airy Road, Basking Ridge, NJ 07920, USA
13. Kundan Singh, Gautam Nair, and Henning Schulzrinne, "Centralized Conferencing using SIP", in Internet Telephony Workshop, New York, April 2001
14. Stefan Berger, Henning Schulzrinne, Stylianos Sidiroglou, and Xiaotao Wu, "Ubiquitous Computing Using SIP", in ACM NOSSDAV 2003, Monterey, California, USC, June 2003
15. Gonzalo Camarillo, Henning Schulzrinne, and Raimo Kantola, "Signalling Transport Protocols", Dept. of Computer Science, Columbia University, New York, Technical report, February 2002
16. Petri Koskelainen, Henning Schulzrinne, and Xiaotao Wu, "A SIP-based Conference Control Framework", in Proc, International Workshop on Network and Operating System Support for Digital Audio and Video(NOSSDAV), Miami Beach, Florida, May 2002
17. Jonathan Lennox and Henning Schulzrinne, "Feature Interaction in Internet Telephony", in Feature Interaction in Telecommunications and Software Systems VI, Glasgow, United Kingdom, May 2000
18. Jonathan Lennox and Henning Schulzrinne, "A Protocol for Reliable Decentralized Conferencing", in ACM NOSSDAV 2003, Monterey, California, USC, June 2003
19. Henning Schulzrinne and Knarig Arabshian, "Providing Emergency Services in Internet Telephony", IEEE Internet Computing, vol. 6, pp. 39-47, May 2002
20. Henning Schulzrinne and J. Rosenberg, "Internet Telephony: Architecture and Protocols -- an IETF perspective," Computer Networks and ISDN Systems, vol. 31, no. 3, pp. 237-255, February 1999

Documents

1. SIP Tutorial : SIP에 관한 다양한 Protocol 설명과 관련 내용
2. SIP Products : SIP를 이용한 Product를 만든 곳에 대한 정보
3. Skype Conferencing Test : Skype에서 conferencing 을 사용해보고 쓴 문서
4. VoIP and Skype Security : General VoIP에 대한 소개와 Skype와 다른 application에 대한 비교 (by Simson L. Garfinkel)
5. Multimedia Over IP : RTP, RTCP, SIP, RSTP에 대한 설명한 파워포인트 문서
6. Realtime Audio and Video : Realtime Audio와 Video를 다루는 기술에 대한 문서
7. SIP Cartoon : SIP 서비스에 대한 간략한 만화

Links

SIP

Session Initiation Protocol(SIP) Charter ( www.ietf.org/html.charters/sip-charter.html )

SER(SIP Express Router) ( www.iptel.org )

Asterisk ( www.asterisk.org )

VOCAL ( www.vovida.org )

SIP Wiki ( www.toyz.org/cgi-bin/sipwiki.cgi )

SIP Forum ( www.sipforum.org )

MPEG4IP - Open Streaming Video and Audio ( www.mpeg4ip.net )

SIP.edu : Voice Over IP Working Group ( voip.internet2.edu/SIP.edu )

Open Directory : SIP products ( dmoz.org/Computers/Internet/Protocols/SIP/ )

VoIP

voip-info.org ( www.voip-info.org )

VoIP Howto ( www.tldp.org/HOWTO/VoIP-HOWTO.html )

VoIP forum( www.voip-forum.or.kr )

Telephony(VoIP Software)

Skype

Brekeke - VoIP software ( www.brekeke.com )

SIPRequest ( www.siprequest.com )

RTP, RTSP, RTCP

About RTP and the Audio-Video Transport Working Group ( www.cs.columbia.edu/~hgs/rtp )

RTP, Real-time Transport Protocol ( www.networksorcery.com/enp/protocol/rtp.htm )

rtsp.org : Real Time Streaming Protocol(RTSP) ( www.rtsp.org )

Henning Schulzrinne ( www.cs.columbia.edu/~hgs/ )

Network Protocol Guide, Network Monitoring & Analysis Tools ( www.javvin.com )

RFC

Internet RFC/FYI/STD/BCP Archives ( www.faqs.org/rfcs/ )

Network Sorcery ( www.networksorcery.com )

Private

2000년 Multimedia conferencing system 과제

Distributed Processing and Network Management Laboratory

Skype traffic analysis page

Columbia IRT Lab