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SIP APIs for voice and video communications on the web

Published:01 August 2011Publication History

ABSTRACT

Existing standard protocols for the web and Internet telephony fail to deliver real-time interactive communication from within a web browser. In particular, the client-server web protocol over reliable TCP is not always suitable for end-to-end low latency media path needed for interactive voice and video communication. To solve this, we compare the available platform options using the existing technologies such as modifying the web programming language and protocol, using an existing web browser plugin, and a separate host resident application that the web browser can talk to. We argue that using a separate application as an adaptor is a promising short term as well as long-term strategy for voice and video communications on the web.

Our project aims at developing the open technology and sample implementations for web-based real-time voice and video communication applications. We describe the architecture of our project including (1) a RESTful web communication API over HTTP inspired by SIP message flows, (2) a web-friendly set of metadata for session description, and (3) an UDP-based end-to-end media path. All other telephony functions reside in the web application itself and/or in web feature servers. The adaptor approach allows us to easily add new voice and video codecs and NAT traversal technologies such as Host Identity Protocol. We want to make web-based communication accessible to millions of web developers, maximize the end user experience and security, and preserve the huge global investment in and experience from SIP systems while adhering to web standards and development tools as much as possible. We have created an open source prototype that allows you to freely use the conference application by directing a browser to the conference URL.

References

  1. Amirante, A., et al. NTRULO: A tunneling architecture for multimedia conferencing over IP. NEW2AN'10, St. Petersburg, Russia. pp 460--472. Aug. 2010 Google ScholarGoogle ScholarDigital LibraryDigital Library
  2. Fielding, R., et al. Hypertext transfer protocol -- HTTP/1.1. RFC 2616. IETF. Jun.1999 Google ScholarGoogle ScholarDigital LibraryDigital Library
  3. Komu, M., et al. Basic HIP extensions for traversal of NAT. RFC 5770. IETF. Apr.2010Google ScholarGoogle Scholar
  4. Project: voice and video on web. Illinois Institute of Technology. https://sites.google.com/site/vvowproject/Google ScholarGoogle Scholar
  5. Project: Flash based audio and video communication. http://code.google.com/p/flash-videoio/Google ScholarGoogle Scholar
  6. Richardson, L., Ruby, S. RESTful Web Services. O'Reilly. May 2007. ISBN 978-0-596-52926-0 Google ScholarGoogle ScholarDigital LibraryDigital Library
  7. Rosenberg, J., Schulzrinne, H., et al. SIP: session initiation protocol. RFC 3261. IETF. Jun.2002 Google ScholarGoogle ScholarDigital LibraryDigital Library
  8. RTC-Web IETF working charter proposal. Mar.2011. http://rtc-web.alvestrand.com/ietf-activityGoogle ScholarGoogle Scholar
  9. Singh, K., Schulzrinne, H. SIPpeer: a SIP-based P2P Internet telephony client adaptor. Implementation Report. Columbia University. New York, NY. 2004Google ScholarGoogle Scholar
  10. Sinnreich, H., Johnston, A. SIP APIs for communications on the web. IETF Internet draft. "work in progress". Jun 2010Google ScholarGoogle Scholar
  11. Sinnreich, H., Johnston, A., Shim, E., Singh, K. Simple SIP usage scenario for applications in the endpoints. RFC 5638. IETF. Sep.2009Google ScholarGoogle Scholar
  12. Web real-time communications working group charter. W3C. Dec.2010. http://www.w3.org/2010/12/webrtc-charter.htmlGoogle ScholarGoogle Scholar
  13. Fette, I., The websocket protocol, IETF Internet draft, "work in progress". Jun 2011.Google ScholarGoogle Scholar

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  • Published in

    cover image ACM Other conferences
    IPTcomm '11: Proceedings of the 5th International Conference on Principles, Systems and Applications of IP Telecommunications
    August 2011
    111 pages
    ISBN:9781450309752
    DOI:10.1145/2124436

    Copyright © 2011 ACM

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    Association for Computing Machinery

    New York, NY, United States

    Publication History

    • Published: 1 August 2011

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