Multimedia congestion control: circuit breakers for unicast RTP sessions

The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows. This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms

Congestion-Aware Scalable Video Streaming

Postprin

Congestion Control using FEC for Conversational Multimedia Communication

In this paper, we propose a new rate control algorithm for conversational multimedia flows. In our approach, along with Real-time Transport Protocol (RTP) media packets, we propose sending redundant packets to probe for available bandwidth. These redundant packets are Forward Error Correction (FEC) encoded RTP packets. A straightforward interpretation is that if no losses occur, the sender can increase the sending rate to include the FEC bit rate, and in the case of losses due to congestion the redundant packets help in recovering the lost packets. We also show that by varying the FEC bit rate, the sender is able to conservatively or aggressively probe for available bandwidth. We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network simulator and in the real-world and compare it to other congestion control algorithms

Exploration of RTP circuit breaker with applications to video streaming.

Live multimedia streaming is becoming one of the dominant sources of internet traffic, much of which is sent over best-effort networks, i.e. along paths with a wide variety of characteristics. The multimedia traffic should be transmitted using a robust and effective congestion control mechanism to protect the network from congestion collapse. The RTP Circuit Breaker (RTP-CB) is a candidate solution that causes a sender to cease transmission when RTCP message feedback indicates excessive congestion. This paper studies RTP/UDP video traffic and the impact of its bursty behaviour on the network. It considers the potential limitations of using a RTP-CB with video traffic. We found that the bursty nature of a typical video flow can cause the RTP-CB to either prematurely cease transmission or to react too late. To reduce the likelihood of this happening, we suggest the use of a smoothing buffer in conjunction with the RTP-CB and propose design criteria for this buffer. Our experiments confirm the effectiveness of the proposed approach for different video streams

Real-time Audio-Visual Media Transport over QUIC

We consider the problem of how to transport low-latency, interactive, real-time traffic over QUIC. This is needed to support applications like WebRTC, but difficult to support due to the reliable, unframed, nature of QUIC streams. We review the needs of low-latency real-time applications and how they have been supported in previous protocols, then propose a minimal set of extensions to QUIC to provide such support. Compared to a raw datagram service, our extensions provide meaningful support for partially reliable and real-time flows, in a backwards compatible manner

Media usability circuit breakers for RTP-based interactive networked multimedia

Abstract — With multimedia and Internet enabled devices being ubiquitous, mechanisms that ensure multimedia flows do not congest the Internet are crucial components of multimedia systems that are embraced rather than opposed by network service providers. The Real-time Transport Protocol (RTP) Circuit Breaker is designed to terminate RTP/UDP flows that cause excessive congestion in the network. Multimedia users congesting the network have their flows terminated, as dictated by the RTP circuit breaker congestion rule. Users who obtain little quality from a multimedia session, and consume network resources to no avail, should also cease transmission. This is the mandate of the RTP circuit breaker media usability rule. We propose an algorithm for this rule, and show that it avoids wasting network resources on flows that deliver no quality to the user. Index Terms — RTP, Interactive multimedia traffic, Circuit Breaker, WebRT

Protocols and Algorithms for Adaptive Multimedia Systems

The deployment of WebRTC and telepresence systems is going to start a wide-scale adoption of high quality real-time communication. Delivering high quality video usually corresponds to an increase in required network capacity and also requires an assurance of network stability. A real-time multimedia application that uses the Real-time Transport Protocol (RTP) over UDP needs to implement congestion control since UDP does not implement any such mechanism. This thesis is about enabling congestion control for real-time communication, and deploying it on the public Internet containing a mixture of wired and wireless links. A congestion control algorithm relies on congestion cues, such as RTT and loss. Hence, in this thesis, we first propose a framework for classifying congestion cues. We classify the congestion cues as a combination of: where they are measured or observed? And, how is the sending endpoint notified? For each there are two options, i.e., the cues are either observed and reported by an in-path or by an off-path source, and, the cue is either reported in-band or out-of-band, which results in four combinations. Hence, the framework provides options to look at congestion cues beyond those reported by the receiver. We propose a sender-driven, a receiver-driven and a hybrid congestion control algorithm. The hybrid algorithm relies on both the sender and receiver co-operating to perform congestion control. Lastly, we compare the performance of these different algorithms. We also explore the idea of using capacity notifications from middleboxes (e.g., 3G/LTE base stations) along the path as cues for a congestion control algorithm. Further, we look at the interaction between error-resilience mechanisms and show that FEC can be used in a congestion control algorithm for probing for additional capacity. We propose Multipath RTP (MPRTP), an extension to RTP, which uses multiple paths for either aggregating capacity or for increasing error-resilience. We show that our proposed scheduling algorithm works in diverse scenarios (e.g., 3G and WLAN, 3G and 3G, etc.) with paths with varying latencies. Lastly, we propose a network coverage map service (NCMS), which aggregates throughput measurements from mobile users consuming multimedia services. The NCMS sends notifications to its subscribers about the upcoming network conditions, which take these notifications into account when performing congestion control. In order to test and refine the ideas presented in this thesis, we have implemented most of them in proof-of-concept prototypes, and conducted experiments and simulations to validate our assumptions and gain new insights.

How far are we from WebRTC-1.0? An update on standards and a look at what's next

Real-time communication between browsers has represented an unprecedented standardization effort involving both the IETF and the W3C. These activities have involved both the real-time protocol suite and the application-level JavaScript APIs to be offered to developers in order to allow them to easily implement interoperable real-time multimedia applications in the web. This article sheds light on the current status of standardization, with special focus on the upcoming final release of the so-called WebRTC-1.0 standard ecosystem. It takes stock of the situation with respect to hot topics such as codecs, session description and stream multiplexing. It also briefly discusses how standard bodies are dealing with seamless integration of the initially competing effort known as “Object Real Time Communications.

Low Latency Reliable Data Sharing Mechanism for UAV Swarm Missions

The use of Unmanned Aerial Vehicle (UAV) swarms is increasing in many commercial applications as well as military applications (such as reconnaissance missions, search and rescue missions). Autonomous UAV swarm systems rely on multi-node interhost communication, which is used in coordination for complex tasks. Reliability and low latency in data transfer play an important role in the maintenance of UAV coordination for these tasks. In these applications, the control of UAVs is performed by autonomous software and any failure in data reception may have catastrophic consequences. On the other hand, there are lots of factors that affect communication link performance such as path loss, interference, etc. in communication technology (WIFI, 5G, etc.), transport layer protocol, network topology, and so on. Therefore, the necessity of reliable and low latency data sharing mechanisms among UAVs comes into prominence gradually. This paper examines available middleware solutions, transport layer protocols, and data serialization formats. Based on evaluation results, this research proposes a middleware concept for mobile wireless networks like UAV swarm systems