WebRTC using JSON via XMLHttpRequest and SIP over WebSocket - Initial Signalling Overhead Findings
Michael Adeyeye, Ishmael Makitla, Thomas Fogwill
2013
Abstract
Web Real-Time Communication (WebRTC) introduces real-time multimedia communication as native capabilities of Web browsers. With the adoption of WebRTC the Web browsers will be able to use WebRTC to communicate with one another (peer-to-peer), and with WebSocket servers such as Mobicents SIP Servlets and other server technologies that support WebSocket communication to enable SIP-to-WebRTC communication. This position paper discusses the two common methods of doing real-time communication in Web browsers throughWebRTC. The methods are JavaScript Object Notation (JSON) via XMLHttpRequest (XHR) and Session Initiation Protocol (SIP) via WebSocket. A three-user WebRTC video chat prototype application was developed and used to evaluate both methods. Additional signalling overhead introduced into a browser by each method was determined. The results showed WebRTC-SIP/WS has more overhead than WebRTCJSON/ XHR. This signalling overhead findings are useful in informing the WebRTC working groups in terms of additional overhead introduced by proposed WebRTC methods, the finding could also help application developers make decision on their choice of technologies and protocols when developing WebRTC-supported applications.
References
- WebRTC, http://www.webrtc.org, accessed on October 13, 2011.
- IETF WebRTC, http://tools.ietf.org/wg/rtcweb, accessed on October 13, 2011.
- David Linner, Horst Stein, Ulrich Staiger and Stephan Steglich, “Real-time Communication Enabler for Web 2.0 Applications,” in: Proceedings of the Sixth International Conference on Networking and Services (ICNS 7810), Cancun, Mexico, March 7-13, 2010, pp. 42 - 48.
- SIP on the Web, http://sip-on-the-web.aliax.net, accessed on June 11, 2012.
- SIP-JS, http://code.google.com/p/sip-js, accessed on June 11, 2012.
- The Phono WebRTC, http://phono.com/webrtc, accessed on June 11, 2012.
- Michael Adeyeye, Neco Ventura and Luca Foschini, “Converged Multimedia Services in Emerging Web 2.0 Session Mobility Scenarios ,” in: the Springer Wireless Networks (WINET) Journal. DOI: 10.1007/s11276-011-0394-z.
- Ericsson WebRTC, https://groups.google.com/group/ericssonlabs-web-rtc, accessed on June 11, 2012.
- Chrome WebRTC Implementation, http://www.w3.org/ 2011/04/webrtc/wiki/images/7/7f/Webrtc-chromeimpl-status.pdf, accessed on June 11, 2012.
- IETF RTCWeb-SIP WG, http://tools.ietf.org/html/draftkaplan-rtcweb-sip-interworking-requirements-01, accessed on October 13, 2011.
- The IMS World Forum Summary, http:// www.alanquayle.com/blog/2012/04/the-ims-worldforum-summary-pa.html, accessed on June 11, 2012.
- WebRTC for IE, http://code.google.com/p/webrtc4ie/, Accessed on January 17, 2012.
- SIPML5, http://www.sipml5.org/, accessed on June 11, 2012.
- FF as SIP Endpoint, https://github.com/ethanhugg/ikran, Accessed on January 17, 2012.
- Three User WebRTC Chat Source, https://github.com/ micadeyeye/three-user-webrtc, Accessed on August 3, 2012.
- Vijay K. Gurbani, Xian-He Sun and A. Brusilovsky, “Inhibitors for Ubiquitous Deployment of Services in the Next-Generation Network,” in: the IEEE Communications Magazine, Vol. 43, No. 9, pp. 116-121, September 2005.
- Karim Sbata, Houda Khrouf, Sabine Zander and Monique Becker, “Converging Web and IMS Services: Stakes and Solution Proposals,” in: Proceedings of the International ACM Conference on Management of Emergent Digital EcoSystems (MEDES 7809), Lyon, France, October 27-30, 2009.
- Haruno Kataoka, Masashi Toyama, Yoshiko Sueda, Osamu Mizuno and Kenji Takahashi, “Demonstration of Web Contents Collaborative System for Call Parties,” in: Proceedings of the 7th IEEE Consumer Communications and Networking Conference (IEEE CCNC 7810), Las Vegas, Nevada, USA, January 9-12, 2010.
- Google Chrome Frame, http://www.google.com/ chromeframe?quickenable=true, Accessed on January 17, 2012.
- Google WebRTC Samples, http:// code.google.com/p/ webrtc-samples/, Accessed on January 17, 2012.
- Google AppRTC, https://apprtc.appspot.com, Accessed on January 17, 2012.
- Hideo Nishimura, Hiroyuki Ohnishi and Miki Hirano, “Architecture for Web-IMS Co-operative Services for Web Terminals ,” in: Proceedings of the 13th International Conference on Intelligence in Next Generation Networks (ICIN 7809), Bordeaux, France, October 26 - 29, 2009, pp 1-6.
- The PJSIP Project, http://www.pjsip.org, April 12, 2012.
- The Mozilla Firefox Web browser, http://www.mozilla.org, April 12, 2012.
- M. Handley and V. Jacobson, “SDP: Session Description Protocol,” IETF RFC 2327, April 12, 2012.
Paper Citation
in Harvard Style
Adeyeye M., Makitla I. and Fogwill T. (2013). WebRTC using JSON via XMLHttpRequest and SIP over WebSocket - Initial Signalling Overhead Findings . In Proceedings of the 9th International Conference on Web Information Systems and Technologies - Volume 1: WEBIST, ISBN 978-989-8565-54-9, pages 119-124. DOI: 10.5220/0004317901190124
in Bibtex Style
@conference{webist13,
author={Michael Adeyeye and Ishmael Makitla and Thomas Fogwill},
title={WebRTC using JSON via XMLHttpRequest and SIP over WebSocket - Initial Signalling Overhead Findings},
booktitle={Proceedings of the 9th International Conference on Web Information Systems and Technologies - Volume 1: WEBIST,},
year={2013},
pages={119-124},
publisher={SciTePress},
organization={INSTICC},
doi={10.5220/0004317901190124},
isbn={978-989-8565-54-9},
}
in EndNote Style
TY - CONF
JO - Proceedings of the 9th International Conference on Web Information Systems and Technologies - Volume 1: WEBIST,
TI - WebRTC using JSON via XMLHttpRequest and SIP over WebSocket - Initial Signalling Overhead Findings
SN - 978-989-8565-54-9
AU - Adeyeye M.
AU - Makitla I.
AU - Fogwill T.
PY - 2013
SP - 119
EP - 124
DO - 10.5220/0004317901190124