Cloud gaming is a new class of services that promises to revolutionize the videogame market. It allows the user to play a videogame with basic equipment while using a remote server for the actual execution. The multimedia content is streamed through the network from the server to the user. This service requires low latency and a large bandwidth to work properly with low response time and high-definition video. Three of the leading tech companies, (Google, Sony and NVIDIA) entered this market with their own products, and others, like Microsoft and Amazon, are planning to launch their own platforms in the near future. However, these companies released so far little information about their cloud gaming operation and how they utilize the network. In this work, we study these new cloud gaming services from the network point of view. We collect more than 200 packet traces under different application settings and network conditions for 3 cloud gaming services, namely Stadia from Google, GeForce Now from NVIDIA and PS Now from Sony. We analyze the employed protocols and the workload they impose on the network. We find that GeForce Now and Stadia use the RTP protocol to stream the multimedia content, with the latter relying on the standard WebRTC APIs. They result in bandwidth-hungry and consume up to 45 Mbit/s, depending on the network and video quality. PS Now instead uses only undocumented protocols and never exceeds 13 Mbit/s.

pdf/a-network-analysis-on-cloud-gaming-stadia-geforce-now-and-1yuxyd8vfn.pdf

A network analysis on cloud gaming: Stadia, GeForce Now and PSNow

Most multimedia group and inter-stream synchronization techniques define or use proprietary protocols with new control messages. Many multimedia applications have been developed using RTP/RTCP as the standard for transmission of multimedia streams over IP networks. Instead of defining a new protocol, we propose the use of RTP/RTCP to provide synchronization. We take advantage of the feedback capabilities provided by RTCP and the ability to extend the protocol by extending and creating RTCP messages containing synchronization information. We have implemented our proposal and tested it in our University WAN. Our experiments have shown that network load resulting from synchronization is minimized and that asynchronies are within acceptable limits for multimedia applications.

An RTP/RTCP based approach for multimedia group and inter-stream synchronization

Real-time multimedia flows comprise a large, and increasing, fraction of the traffic on the Internet. An important subset of that traffic, primarily due to interactive applications, runs over UDP/IP, and requires applications to implement congestion control to ensure the stability of the network. The IETF is developing congestion control algorithms for such uses as part of the new WebRTC standards, but there is no standard algorithm that can be used at this time. We do not propose a congestion control algorithm. Rather, we propose a circuit breaker for RTP sessions that can detect when an application is causing excessive network congestion, and shut down the transmission. This can be used as an envelope within which congestion control algorithms can operate, providing a safety net to prevent congestion collapse. We present the RTP circuit breaker algorithm, and provide an initial performance evaluation to show that it performs as desired.

https://csperkins.org/publications/2013/12/singh2013circuitbreaker.pdf

Circuit Breakers for Multimedia Congestion Control

QUIC is the new transport protocol for the Internet, designed for secure, reliable communication, especially with the web in mind. While multimedia streaming, allowing for some playout delay, has been widely run on top of reliable transport protocols, conversational multimedia usually requires unreliable ones, often using RTP over UDP. A recent extension to QUIC supports unreliable datagrams within QUIC connections so that also real-time media can be supported. In this paper, we investigate a recent design for RTP over QUIC with a focus on congestion control and the related signaling. We implement a strawman using Gstreamer and quic-go and evaluate different permutations of congestion control algorithms and signaling in a simple testbed.

Congestion control for real-time media over QUIC

This paper proposes a video streaming system optimizing resource utilization when the media server only disposes of long term feedbacks from the client. Based on a partial knowledge of the network, we developed a scheduling algorithm that exploits the scalable video coding (SVC) properties to estimate packets importance and that takes into account packet delay dependencies to better anticipate congestion situations. Compared to more conventional streaming systems, experimental results show that our approach allows to better face network condition degradation like bandwidth reduction or packet error rate increase.

Adaptive video streaming with long term feedbacks

This document describes the results achieved when simulating the timing rules of the Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback, denoted AVPF. Unicast and multicast topologies are considered as well as several protocol and environment configurations. The results show that the timing rules result in better performance regarding feedback delay and still preserve the well-accepted RTP rules regarding allowed bit rates for control traffic.

/pdf/extended-rtp-profile-for-real-time-transport-control-1pf99g03ch.pdf

Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback: Results of the Timing Rule Simulations

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Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback: Results of the Timing Rule Simulations