Search

This work studies the jitter buffer management algorithm for Voice over IP in WebRTC. In particular, it details the core concepts of WebRTC’s jitter buffer management. Furthermore, it investigates how jitter buffer management algorithm behaves under network conditions with packet bursts. It also proposes an approach, different from the default WebRTC algorithm, to avoid distortions that occur under such network conditions. Under packet bursts, when the packet buffer becomes full, the WebRTC jitter buffer algorithm may discard all the packets in the buffer to make room for incoming packets. The proposed approach offers a novel strategy to minimize the number of packets discarded in the presence of packet bursts. Therefore, voice quality as perceived by the user is improved. ITU-T Rec. P.863, which also confirms the improvement, is employed to objectively evaluate the listening quality.

Nowadays, media content can be delivered via diverse broadband and broadcast technologies. Although these different technologies have somehow become rivals, their coordinated usage and convergence, by leveraging of their strengths and complementary characteristics, can bring many benefits to both operators and customers. For example, broadcast TV content can be augmented by on-demand broadband media content to provide enriched and personalized services, such as multi-view TV, audio language selection, and inclusion of real-time web feeds. A piece of evidence is the recent Hybrid Broadcast Broadband TV (HbbTV) standard, which aims at harmonizing the delivery and consumption of (hybrid) broadcast and broadband TV content. A key challenge in these emerging scenarios is the synchronization between the involved media streams, which can be originated by the same or different sources, and delivered via the same or different technologies. To enable synchronized (hybrid) media delivery services, some mechanisms providing timelines at the source side are necessary to accurately time align the involved media streams at the receiver-side. This paper provides a comprehensive review of how clock references (timing) and timestamps (time) are conveyed and interpreted when using the most widespread delivery technologies, such as DVB, RTP/RTCP and MPEG standards (e.g., MPEG-2, MPEG-4, MPEG-DASH, and MMT). It is particularly focused on the format, resolution, frequency, and the position within the bitstream of the fields conveying timing information, as well as on the involved components and packetization aspects. Finally, it provides a survey of proofs of concepts making use of these synchronization related mechanisms. This complete and thorough source of information can be very useful for scholars and practitioners interested in media services with synchronization demands., This work has been funded, partially, by the "Fondo Europeo de Desarrollo Regional" (FEDER) and the Spanish Ministry of Economy and Competitiveness, under its R&D&i Support Program in project with ref TEC2013-45492-R.

The Internet has evolved to a stage where the number of Internet enabled devices exceeds the global population. With much of the required connectivity over wireless, contention for bandwidth is an important issue that has to be addressed. In this context, applications that have certain quality of service QoS requirements must be protected. WiFi-enabled mobile devices such as smartphones and tablets are commonly used by both personal and business users for voice over IP VoIP communications, while also supporting conventional data applications such as email, file transfer and web access. Previous research has shown that time synchronized endpoints can provide better QoS by calculating accurate mouth-to-ear delays, and using this information to better inform buffer playout strategies. The IEEE 802.11e protocol extends 802.11 by providing different traffic priorities based on traffic type. However, it cannot distinguish between traffic streams within the same category. In this paper, we present an access point-centred mechanism for further distinguishing between streams within 802.11e categories. The mechanism predicts the VoIP quality for multiple sessions using the ITU-T E-Model. Delay values are determined from RTCP packet timing information. We provide implementation details on a real world proof-of-concept and present results that correlate well with previous NS-3 based simulations. In addition, we address scalability issues for large scale deployments. Copyright © 2015 John Wiley & Sons, Ltd.

Various approaches seek to optimize the quality of service of VoIP applications. We propose a system that uses synchronized time to combine the useful characteristics of both fixed and adaptive buffer strategies, thereby improving VoIP quality of service. Using a combination of global positioning system (GPS) technologies and the network time protocol (NTP), hosts can learn the precise end-to-end delay for each packet. This information can benefit both domestic and business Internet telephony users. We outline our proposed system and discuss issues arising from the use of synchronized time.