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QoS analysis for WebRTC videoconference on bandwidth-limited network
Bandung Y.a, Subekti L.B.a, Tanjung D.a, Chrysostomou C.b
a Institut Teknologi Bandung, Indonesia
b Frederick University of Cyprus, Cyprus
[vc_row][vc_column][vc_row_inner][vc_column_inner][vc_separator css=”.vc_custom_1624529070653{padding-top: 30px !important;padding-bottom: 30px !important;}”][/vc_column_inner][/vc_row_inner][vc_row_inner layout=”boxed”][vc_column_inner width=”3/4″ css=”.vc_custom_1624695412187{border-right-width: 1px !important;border-right-color: #dddddd !important;border-right-style: solid !important;border-radius: 1px !important;}”][vc_empty_space][megatron_heading title=”Abstract” size=”size-sm” text_align=”text-left”][vc_column_text]© 2017 IEEE.This paper presents a quality of service analysis conducted for real-time multimedia communication services particularly over limited capacity network. This research aims to examine quality of videoconference service based on WebRTC technology. We investigate several implementation challenges of videoconferencing in bandwidth limited network by conducting a thorough experimental study. We analyze videoconference quality from the implementation of WebRTC based system for supporting distance learning between two schools in Cianjur, West Java, Indonesia. Further, we study adaptability feature of WebRTC by performing experiments for several network conditions. Result of the experiment shows that fair quality of video (peak signal-to-noise ratio, PSNR, between 30dB to 33dB) can be reached when bandwidth capacity is 2048kbps. On the other hand, the experiment shows acceptable audio and video stream quality with packet loss rate (PLR) less than 1% and jitter less than 100ms can be reached in network condition with bandwidth capacity higher than 128kbps. Outcome of this research supports understanding and addressing of the challenges of real-time videoconferencing on bandwidth limited network.[/vc_column_text][vc_empty_space][vc_separator css=”.vc_custom_1624528584150{padding-top: 25px !important;padding-bottom: 25px !important;}”][vc_empty_space][megatron_heading title=”Author keywords” size=”size-sm” text_align=”text-left”][vc_column_text]Audio and video,Bandwidth capacity,Bandwidth limiteds,Network condition,Peak signal to noise ratio,Real-time multimedia communication,Videoconference,WebRTC[/vc_column_text][vc_empty_space][vc_separator css=”.vc_custom_1624528584150{padding-top: 25px !important;padding-bottom: 25px !important;}”][vc_empty_space][megatron_heading title=”Indexed keywords” size=”size-sm” text_align=”text-left”][vc_column_text]Audio and video quality,Bandwidth limited network,Videoconference,WebRTC[/vc_column_text][vc_empty_space][vc_separator css=”.vc_custom_1624528584150{padding-top: 25px !important;padding-bottom: 25px !important;}”][vc_empty_space][megatron_heading title=”Funding details” size=”size-sm” text_align=”text-left”][vc_column_text][/vc_column_text][vc_empty_space][vc_separator css=”.vc_custom_1624528584150{padding-top: 25px !important;padding-bottom: 25px !important;}”][vc_empty_space][megatron_heading title=”DOI” size=”size-sm” text_align=”text-left”][vc_column_text]https://doi.org/10.1109/WPMC.2017.8301873[/vc_column_text][/vc_column_inner][vc_column_inner width=”1/4″][vc_column_text]Widget Plumx[/vc_column_text][/vc_column_inner][/vc_row_inner][/vc_column][/vc_row][vc_row][vc_column][vc_separator css=”.vc_custom_1624528584150{padding-top: 25px !important;padding-bottom: 25px !important;}”][/vc_column][/vc_row]