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Janus: a general purpose WebRTC gateway

Published: 01 October 2014 Publication History

Abstract

This paper deals with the design and implementation of Janus, a general purpose, open source WebRTC gateway. Details will be provided on the architectural choices we took for Janus, as well as on the APIs we made available to extend and make use of it. Examples of how the gateway can be used for complex WebRTC applications are presented, together with some experimental results we collected during the development process.

References

[1]
Doubango webrtc2sip. http://webrtc2sip.org/.
[2]
Janus on GitHub. https://github.com/meetecho/janus-gateway.
[3]
Janus Online Demos. http://janus.conf.meetecho.com.
[4]
Jitsi Videobridge. https://jitsi.org/Projects/JitsiVideobridge.
[5]
Kurento. http://www.kurento.org/.
[6]
Licode. http://lynckia.com/licode/.
[7]
Medooze. http://www.medooze.com/.
[8]
Real-Time Communication in WEB-browsers (IETF). http://tools.ietf.org/wg/rtcweb/charters.
[9]
Selenium Framework. http://docs.seleniumhq.org/.
[10]
Tiscali Indoona. http://www.indoona.com/.
[11]
Web Real-Time Communications Working Group Charter (W3C). http://www.w3.org/2011/04/webrtc-charter.html.
[12]
A. Amirante, T. Castaldi, L. Miniero, and S. P. Romano. Separation of Responsibilities between Application Servers and Media Servers in NGNs: A Practical Approach. In Next Generation Teletraffic and Wired/Wireless Advanced Networking, pages 199--211. Springer, 2008.
[13]
M. Baugher, D. McGrew, and M. N. et al. The Secure Real-time Transport Protocol (SRTP). RFC 3711, RFC Editor, March 2004.
[14]
C. Holmberg, H. Alvestrand, and C. Jennings. Negotiating Media Multiplexing Using the Session Description Protocol (SDP). Internet-Draft draft-ietf-mmusic-sdp-bundle-negotiation-08, IETF Secretariat, Apr. 2014.
[15]
E. Ivov, E. Rescorla, and J. Uberti. Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol. Internet-Draft draft-ietf-mmusic-trickle-ice-01, IETF Secretariat, Feb. 2014.
[16]
R. Jesup, S. Loreto, and M. Tuexen. WebRTC Data Channels. Internet-Draft draft-ietf-rtcweb-data-channel-11, IETF Secretariat, June 2014.
[17]
S. Ludwig, J. Beda, and P. Saint-Andre. XEP-0166: Jingle. Technical report, XMPP Standards Foundation, December 2009.
[18]
D. McGrew and E. Rescorla. Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP). RFC 5764, RFC Editor, May 2010.
[19]
L. Miniero, V. Pascual, and S. G. Murillo. Guidelines to support RTCP end-to-end in Back-to-Back User Agents (B2BUAs). Internet-Draft draft-ietf-straw-b2bua-rtcp-01, IETF Secretariat, Dec. 2013.
[20]
C. Perkins, M. Westerlund, and J. Ott. Web Real-Time Communication (WebRTC): Media Transport and Use of RTP. Internet-Draft draft-ietf-rtcweb-rtp-usage-17, IETF Secretariat, Apr. 2014.
[21]
J. Rosenberg. Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols. RFC 5245, RFC Editor, April 2010.
[22]
J. Rosenberg and H. Schulzrinne. An Offer/Answer Model with the Session Description Protocol (SDP). RFC 3264, RFC Editor, June 2002.
[23]
J. Rosenberg, H. Schulzrinne, and G. C. et al. SIP: Session Initiation Protocol. RFC 3261, RFC Editor, June 2002.
[24]
H. Schulzrinne, S. Casner, and R. F. et al. RTP: A Transport Protocol for Real-Time Applications. RFC 3550, RFC Editor, July 2003.
[25]
J. Uberti and C. Jennings. Javascript Session Establishment Protocol. Internet-Draft draft-ietf-rtcweb-jsep-07, IETF Secretariat, Feb. 2014.

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      IPTComm '14: Proceedings of the Conference on Principles, Systems and Applications of IP Telecommunications
      October 2014
      65 pages
      ISBN:9781450321242
      DOI:10.1145/2670386
      Permission to make digital or hard copies of all or part of this work for personal or classroom use is granted without fee provided that copies are not made or distributed for profit or commercial advantage and that copies bear this notice and the full citation on the first page. Copyrights for components of this work owned by others than the author(s) must be honored. Abstracting with credit is permitted. To copy otherwise, or republish, to post on servers or to redistribute to lists, requires prior specific permission and/or a fee. Request permissions from [email protected].

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      Published: 01 October 2014

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      Author Tags

      1. IETF
      2. JavaScript
      3. Rtcweb
      4. WebRTC
      5. conferencing
      6. gateways
      7. streaming
      8. web communication

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