There are four special sessions accepted for PV 2013:
- Low-Latency Interactive Video
- Video Delivery over Information-Centric Networks
- WebRTC 2.0
- Advances in Adaptive Streaming
Note that the submission deadline for these special sessions is the same as the regular paper submission deadline. When submitting your paper, choose the track appropriately.
Several years ago, it was found that users do not like video quality fluctuations. At that time the predominant belief was that network rate fluctuations should be minimized, in order to reasonably interoperate with TCP in the network. This led to the creation of a number of so-called “TCP-friendly” congestion controls that exhibit a smoother sending rate than TCP, while avoiding sending more than a conformant TCP under similar conditions. TFRC is perhaps the best known example of such a congestion control mechanism.
A lot has happened since then:
- The notion of TCP-friendliness has received massive criticism; the widespread deployment of a more aggressive TCP variant, CUBIC, has not led to an Internet meltdown, making the case that diverging from strict TCP-friendliness is possible.
- Latency minimization has become a major goal, especially in the face of “bufferbloat”: large delays from large buffers with simplistic FIFO-queue management.
- Codecs have improved; novel video codecs are able to adjust the data rate, but modern codecs may also produce variable bitrate transmissions with burstier packet flows than before.
- TFRC has been embedded in the DCCP protocol, which has probably never been used for anything other than experiments; instead of running over DCCP, RTP-based applications now contain proprietary congestion control mechanisms.
The emergence of the RTCWEB protocol suite for real-time communication between Web browsers has renewed the interest in developing congestion control standards for real-time media. This time, however, the goal is to get things right: delay should be minimized, and standards should realize congestion control using RTP with RTCP signaling. The IETF “Real-time Media Congestion Avoidance Techniques” (RMCAT) working group has been founded to address this need. New questions arise: what type of congestion controls do we need? How much feedback should we send? How do we make this work in multi-user scenarios, e.g., for video conferencing? What should be the API between a video codec and a new delay-based congestion controlled RTP stream? What is the quality that can be expected from the combination of a codec and congestion control mechanism, when we consider better metrics than plain PSNR?
Topics of interest include, but are not limited to:
- Congestion control algorithms for interactive real-time video: requirements, evaluation criteria, and mechanisms
- Necessary RTP/RTCP extensions
- Field experience with video codecs in a low-delay, real-time setting
- Interactions between applications and RTP flows
- Failing to meet real-time schedules: impact, techniques to detect, instrument or diagnose it
This activity is partially funded by the European Community under its Seventh Framework Programme through the Reducing Internet Transport Latency (RITE) project (ICT-317700).
- Michael Welzl, University of Oslo (michawe at ifi.uio.no)
- Stein Gjessing, University of Oslo (steing at ifi.uio.no)
Information-centric networking (ICN) is a new networking field that recasts content delivery via packet networks. Instead of host-centric routing of packets based on addressed endpoints, ICN raises named data to first-class citizen status and routes directly based on data names. The role of the network is to find and deliver named and secured chunks of data to consumers who request the data by name, independent of the actual source. This introduces fundamental changes in transport protocols, routing, caching, and network behavior. The field of ICN has seen significant research advances in recent years, with multiple projects worldwide attempting to solve the problems of designing a new ground-up networking system. ICN’s focus on content delivery offers special opportunities for video applications; however, these opportunities come with new challenges as well. This session will focus on video delivery and streaming over ICN networks, including identification of key challenges and opportunities for both existing video applications and new applications and delivery techniques enabled by ICN.
- Ashok Narayanan, Cisco (ashokn at cisco.com)
- Jeff Burke, UCLA (jburke at remap.ucla.edu)
The standardization of version 1.0 of Web Real-time Communications (WebRTC) is expected to complete by the end of the year; yet many technical and especially research challenges remain open, and we expect new ones to arise with deployment experience. With WebRTC and the accompanying APIs, application developers have the opportunity to add new immersive features (gestures, real-time communication, peer-to-peer) within their web applications. Apart from the multimedia aspect, WebRTC permits sending data packets between the peers using ‘Data Channels’, which opens the door for innovative apps and new research ideas.
Mobile devices are a big market opportunity and challenge, WebRTC is no exception. The wireless environment and mobility in general pose some interesting problems: call setup, mobility, handover, etc. In general, the WebRTC-enabled services and legacy services (SIP/XMPP-enabled, Skype) face the challenge of interoperability, especially since WebRTC does not standardize a signaling protocol. Lastly, Internet Service Providers (ISPs, both mobile and broadband) have to operate in the tension between present and future services. They need to be able to engineer their network in a way supportive for transmitting the WebRTC flows at an unprecedented scale.
The objective of this special session is to bring together researchers and practitioners in the area of real-time communications, multimedia systems, transport protocols, broadband and mobile networks, and multimedia applications to advance the state of research in real-time communication. We solicit original contributions on advanced topics in web-based real-time multimedia communications.
Topics of particular interest include, but are not limited to:
- Mobile WebRTC
- Architectures for media transport, multiplexing and naming
- Identity management and security
- Operations and management
- Understanding QoE and performance of WebRTC
- WebRTC applications
- Deployment issues
- Non-media applications or data channels
- WebRTC and economy of scale
- Varun Singh, Aalto University (varun.singh at aalto.fi)
- Jörg Ott, Aalto University (jorg.ott at aalto.fi)
- Colin Perkins, University of Glasgow (csp at csperkins.org)
There is a significant market move towards adopting adaptive streaming as the method of choice for scalable over-the-top (OTT) streaming of video, of which MPEG Dynamic Adaptive Streaming over HTTP (DASH) and HTTP Live Streaming (HLS) are relevant examples. Even though adaptive streaming has been around in various forms for a few years, its potential and characteristics are not completely realized or understood, and that is the focus of this session.
Examples of possible adaptive streaming topics include:
- Characterization of a good user experience for adaptive streaming
- Aggregate quality of experience provided by multiple competing clients
- Delivery of adaptive streaming content over other transports, such as eMBMS
- Client architectures and algorithms
- Content delivery architectures and algorithms
- Content preparation and ingestion architectures and algorithms
- Algorithms and protocols for delivering advertising
- Enablement of near-real-time streaming
- Interoperability guidelines and profiles
- Michael Luby, Qualcomm (luby at qti.qualcomm.com)