The Quality Comparison of WebRTC and SIP Audio and Video Communications with PSNR

  • Muhhamad Affan Hasby Telkom University
  • Aji Gautama Putrada
  • Febri Dawani
Abstract views: 305 , 67 downloads: 697
Keywords: Video Communication, WebRTC, SIP, PSNR, FreePBX, NodeJS, Video, Audio

Abstract

Video and audio communications have become part of all areas of work. Two real-time
communication protocols commonly used for IP-based video and audio communications
are Session Initiation Protocol (SIP) and real-time web communications (WebRTC). Both
protocols have been widely used in softphone and video conferencing applications. The
main objective of this research is to make an analysis of the performance of a client server
application for video and audio communications developed by SIP and WebRTC. The SIP
system consists of a softphone on the client side using Bria and a FreePBX server, forWebRTC
applications, using JavaScript and a server at Node.js. The results showed that the WebRTC
audio and video communication provided better quality in terms of PSNR. This is due to
the different codecs used between WebRTC and SIP. WebRTC uses VP8 as video codec, SIP
uses H.246 as video codec, WebRTC uses G.711 as audio codec, and implemented SIP uses
G.729 as audio codec.

Downloads

Download data is not yet available.

References

[1] Branislav Sredojev, D. Samardzija, and Dragan Posarac. Webrtc technology overview and signaling solution design and
implementation. 2015 38th International Convention on Information and Communication Technology, Electronics and
Microelectronics (MIPRO), pages 1006–1009, 2015.
[2] Boris Magnusson, Thomas Sandholm, and Björn Johnsson. An on-demand webrtc and iot device tunneling service for
hospitals. 08 2014.
[3] Salvatore Loreto and Simon Pietro Romano. Real-time communication with WebRTC: peer-to-peer in the browser. " O’Reilly
Media, Inc.", 2014.
[4] K. K. Tam and H. L. Goh. Session initiation protocol. In 2002 IEEE International Conference on Industrial Technology,
2002. IEEE ICIT ’02., volume 2, pages 1310–1314 vol.2, 2002.
[5] G. Suciu, S. Stefanescu, C. Beceanu, and M. Ceaparu. Webrtc role in real-time communication and video conferencing. In
2020 Global Internet of Things Summit (GIoTS), pages 1–6, 2020.
[6] Hardik Tandel and Dr. Parag Rughani. Forensic analysis of asterisk-freepbx based voip server. International Journal of
Emerging Research in Management and Technology, 6:166, 06 2018.
[7] TP Fowdur, N Ramkorun, and PK Chiniah. Performance analysis of webrtc and sip-based audio and video communication
systems. SN Computer Science, 1(6):1–22, 2020.
[8] Navrattan Parmar and Virender Ranga. Performance analysis of webrtc and sip for video conferencing. Int. J. Innov. Technol.
Explor. Eng.(IJITEE), 8(9S):679–686, 2019.
[9] Li Yan. Research and design of rich media communication system based on webrtc and sip. Master’s thesis, Xi’an University
of Science and Technology, 2018.
[10] Pavel Segeˇc, Peter Palúch, Jozef Papán, and Milan Kubina. The integration of webrtc and sip: way of enhancing realtime,
interactive multimedia communication. In 2014 IEEE 12th IEEE International Conference on Emerging eLearning
Technologies and Applications (ICETA), pages 437–442. IEEE, 2014.
[11] Matthew Hoy. Html5: A new standard for the web. Medical reference services quarterly, 30:50–5, 01 2011.
[12] R. Mahy and P. Matthews. Session traversal utilities for nat (stun). 5389, 01 2008.
[13] J. Rosenberg. Traversal using relays around nat (turn): Relay extensions to session traversal utilities for nat (stun). 01 2010.
[14] Alan B Johnston. SIP: understanding the session initiation protocol. Artech House, 2015.
[15] Eric Coll and M Eng. Telecom 101 Telecommunications Reference Book: 2016 Fourth Edition. High-Quality Reference Book
and Study Guide Covering All Major Telecommunications Topics... in Plain English. Teracom Training Institute, 2016.
[16] Quan Huynh-Thu and Mohammed Ghanbari. Scope of validity of psnr in image/video quality assessment. Electronics letters,
44(13):800–801, 2008.
[17] Yousef Sharrab and Nabil Sarhan. Detailed comparative analysis of vp8 and h.264. pages 133–140, 12 2012.
[18] Daniel C Burnett. Webrtc: Handling media on the web. In Multimodal Interaction with W3C Standards, pages 155–169.
Springer, 2017.
[19] Adauto Cavalcante Menezes, Toniclay Andrade Nogueira, Edward David Moreno Ordonez, and Admilson de Ribamar Lima
Ribeiro. An approach to the performance and efficiency power analysis on embedded devices using asterisk. 2018.
[20] Gaous Afrizal et al. Impact of random and burst packet loss on voice codec g. 711, g. 722, g. 729, amr-nb, amr-wb. In 2018
4th International Conference on Wireless and Telematics (ICWT), pages 1–4. IEEE, 2018.
Published
2021-05-03
How to Cite
Muhhamad Affan Hasby, Putrada, A. G., & Dawani, F. (2021). The Quality Comparison of WebRTC and SIP Audio and Video Communications with PSNR. Indonesia Journal on Computing (Indo-JC), 6(1), 73-84. https://doi.org/10.34818/INDOJC.2021.6.1.549
Section
Computer Science

Most read articles by the same author(s)