Dynamic video bitrate adaptation for WebRTC-based remote teaching applications

Abstract

Abstract-Remote teaching applications are common nowadays. Very often, these applications resemble video-on-demand streaming platforms rather than real virtual classrooms, where a group of students (the receivers) can remotely attend a live lecture held by a lecturer (the sender). To better support this live scenario, Real-Time Communication (RTC) solutions can be used. WebRTC is an open-source project for real-time browserbased conferencing, developed with a peer-to-peer architecture in mind. To use WebRTC, each receiver requires a dedicated encoder at sender-side. Using such approach is expensive in terms of encoders, and does not scale well for a large number of users. To overcome this issue, a WebRTC-compliant framework is proposed, where only a limited number of encoders are used. A centralized node, the conference controller, dynamically forwards the most suitable stream to the receivers, based on their bandwidth conditions. Moreover, the controller dynamically recomputes the encoding bitrates of the sender. This approach allows to closely follow the long-term bandwidth variations of the receivers, even with a limited number of encoders at senderside. To evaluate the performance of the proposed framework in a realistic environment, a testbed has been implemented using the Chrome browser and the open-source Jitsi-Videobridge. In a scenario with 10 receivers and 3 encoders, and under realistic network conditions, the proposed framework improves the received video bitrate up to 11%, compared to a static solution where the encoding bitrates do not change over time.

Improving quality and scalability of webRTC video collaboration applications

“…and therefore maximize the received bitrate. A preliminary evaluation of the proposed framework has been presented in previous work [11], where we showed that the dynamic bitrate recomputation results in 15% higher video rate at the receivers compared to a static, fixed association of the encoding bitrates.…”

Quality and Transferred Data Based Video Bitrate Control Method for Web-Conferencing

“…Bitrate adaptation methods for WebRTC have been studied [10]- [12], [14], [15]. GCC [10], [11], which is one of the well-known methods, controls bitrate on the basis of jitter and packet loss rate.…”

“…Wang et al proposed a scalable video coding (SVC) layer bitrate selection algorithm for multiparty interactive live video streaming to maximize the total quality of the receiver on the basis of the sender's uplink throughput, receiver's downlink throughput, and buffer occupancy [12]. Petrangeli et al proposed a video bitrate control method for WebRTC-based remote teaching applications that maximizes the video bitrate transmitted to the receiver, taking into account the available bandwidth of the receiver [15]. This method can control the video bitrate in response to bandwidth fluctuations.…”

“…This method can control the video bitrate in response to bandwidth fluctuations. Although these methods [10]- [12], [14], [15] can improve the quality, the amount of transferred data cannot be reduced because excessive quality is provided when the network quality is high enough.…”

Quality-based Video Bitrate Control for WebRTC-based Teleconference Services

“…Their method introduces a conference controller and instructed the sender to encode fewer bitrate patterns than the number of receivers to reduce the encoding cost. In addition, it was found that the bandwidth could be efficiently utilized by varying the encoding bitrates in accordance with the bandwidth of the receiver [7]. These methods are expected to reduce server infrastructure costs, but that is not expected to reduce the transferred data volume.…”