• ETRI Journal
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• v.32 no.2
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• pp.281-291
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• 2010

Scalable video coding (SVC) has been standardized as an extension of the H.264/AVC standard. This paper proposes a practical real-time transport protocol (RTP) packetization scheme to transport SVC video over IP networks. In combined scalability of SVC, a coded picture of a base or scalable enhancement layer is produced as one or more video layers consisting of network abstraction layer (NAL) units. The SVC NAL unit header contains a (DID, TID, QID) field to identify the association of each SVC NAL unit with its scalable enhancement layer without parsing the payload part of the SVC NAL unit. In this paper, we utilize the (DID, TID, QID) information to derive hierarchical spatio-temporal relationship of the SVC NAL units. Based on the derivation using the (DID, TID, QID) field, we propose a practical RTP packetization scheme for generating single RTP sessions in unicast and multicast transport of SVC video. The experimental results indicate that the proposed packetization scheme can be efficiently applied to transport SVC video over IP networks with little induced delay, jitter, and computational load.

• ETRI Journal
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• v.30 no.3
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• pp.441-450
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• 2008

The scalable extension of H.264, known as scalable video coding (SVC) has been the main focus of the Joint Video Team's work and was finalized at the end of 2007. Synchronization between media is an important aspect in the design of a scalable video streaming system. This paper proposes an efficient media synchronization mechanism for SVC video transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, a real-time transport protocol/RTP control protocol (RTP/RTCP) suite is usually employed. To provide an efficient mechanism for media synchronization between SVC video and audio, we suggest an efficient RTP packetization mode for inter-layer synchronization within SVC video and propose a computationally efficient RTCP packet processing method for inter-media synchronization. By adopting the computationally simple RTCP packet processing, we do not need to process every RTCP sender report packet for inter-media synchronization. We demonstrate the effectiveness of the proposed mechanism by comparing its performance with that of the conventional method.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.2
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6C
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• pp.471-479
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• 2008

In this paper, we propose a novel algorithm to automatically generate an RTP timestamp value that is required for the RTP packetization in order to transmit SVC video over various If networks such as Internet. Unlike the conventional single layer coding algorithms such as H.263, MPEG-4 and H.264, SVC generates a multi-layered single bitstream which is composed of a base layer and one or more enhancement layers in order to simultaneously provide temporal, spatial, and SNR scalability. Especially, in order to provide temporal scalability based on hierarchical B-picture prediction structure, the encoding (or transmission) and display order of pictures in SVC coding is completely decoupled. Thus, the timestamp value to be specified at the header of each RTP packet in video transmission does not increase monotonically according to the display time instant of each picture. Until now, no method for automatically generating an RTP timestamp when SVC video is loaded in a RTP packet has teen introduced. In this paper, a novel automatic RTP timestamp generation method exploiting the TID (temporal ID) field of the SVC NAL unit header is proposed to accommodate the SVC video transmission.

• Journal of KIISE:Information Networking
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• v.36 no.1
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• pp.34-42
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• 2009

In this paper, we propose a practical hybrid transport error control scheme to provide SVC video streaming service over error-prone IP networks. Many error control mechanisms for various video coding standards have been proposed in the literature. However, there is little research result which can be practically applicable to the multilayered coding structure of SVC(the scalable extension of H.264/AVC). We present a new hybrid transport error control scheme that efficiently combines layered Forward Error Correction(FEC) and Automatic Repeat Request(ARQ) for better packet-loss resilience. In the proposed hybrid error control, we adopt ACK-based ARQ instead of NACK-based ARQ to maximize throughput which is the amount of effective data packets delivered over a physical link per time unit. In order to prove the effectiveness of the proposed hybrid error control scheme, we adopt NIST-Net network emulator which is a general-purpose tool for emulating performance dynamics in IP networks. It is shown by simulations over the NIST-Net that the proposed hybrid error control scheme shows improved packet-loss resilience even with much less number of overhead packets compared to various conventional error control schemes.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.872-883
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• 2013

In this paper, we design and analyze performance of Exclusive-OR based FEC (Forward error correction) system to deploy SVC video transmission service over packet-loss prone IP network. In the designed system, we adopt standard compliant Exclusive-OR based FEC scheme and apply it to be appropriate to the hierarchical layer structure of SVC video. To verify the performance of the designed Exclusive-OR based FEC system for SVC video transmission, we employ NIST-NET based transport simulator. By the SVC video transmission using the NIST-NET based simulator, we confirm the error resilient transmission performance of the designed Exclusive-OR based FEC system.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.8
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• pp.3121-3135
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• 2015

Owing to the increase in consumption of multimedia content and the improvement of device capacity, user demand for high-quality content has increased. However, it is difficult to transport such large amounts of content over the existing broadcasting network with limited bandwidth. To provide high-definition broadcasting, some studies suggest methods of transporting multimedia over heterogeneous networks after encoding content hierarchically. MPEG Media Transport (MMT), standardized by Moving Picture Experts Group (MPEG), is a solution that enables large-volume media transport over heterogeneous networks such as digital broadcasting networks and packet-switched networks. In the case of delivering a scalable encoded video over different networks, synchronization of each stream is an important issue. MMT defines a synchronization scheme, but does not contain sufficient functions to implement it. In this paper, we propose a synchronization scheme for media streams that are encoded hierarchically, divided into layers, and transported over heterogeneous networks. We implement our scheme using MMT and HTTP, and experimented using three encapsulated video streams with different durations. As a result, we show that the proposed scheme can reduce the waiting time to display high-quality video, relative to Dynamic Adaptive Streaming over HTTP-Scalable Video Coding (DASH-SVC) by requesting segments of enhanced layers after calculating the transmission time. Additionally, we find out that the selection of durations have a relation to the characteristics of the video.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.10
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• pp.2430-2447
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• 2013

In recent years, HTTP adaptive streaming (HAS) has attracted considerable attention as the state-of-the-art technology for video transport. HAS dynamically adjusts the quality of video streaming according to the network bandwidth and device capability of users. Content-Centric Networking (CCN) has also emerged as a future Internet architecture, which is a novel communication paradigm that integrates content delivery as a native network primitive. These trends have led to the new research issue of harmonizing HAS with the in-network caching provided by CCN routers. Previous research has shown that the performance of HAS can be improved by using the H.264/SVC(scalable video codec) in the in-network caching environments. However, the previous study did not address the misbehavior that causes video freeze when overestimating the available network bandwidth, which is attributable to the high cache hit rate. Thus, we propose a new SVC-based adaptation algorithm that utilizes a drop timer. Our approach aims to stop the downloading of additional enhancement layers that are not cached in the local CCN routers in a timely manner, thereby preventing excessive consumption of the video buffer. We implemented our algorithm in the SVC-HAS client and deployed a testbed that could run Smooth-Streaming, which is one of the most popular HAS solutions, over CCNx, which is the reference implementation of CCN. Our experimental results showed that the proposed scheme (SLA) could avoid video freeze in an effective manner, but without reducing the high hit rate on the CCN routers or affecting the high video quality on the SVC-HAS client.

• Proceedings of the Korean Information Science Society Conference
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• 2008.06a
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• pp.209-210
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• 2008

• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.520-529
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• 2011

In this paper, we propose the extension of the MPEG-2 Transport Stream (TS) header for efficient adaptation of multi-layer coded video such as scalable video coding (SVC) and multiview video coding (MVC) in the HTTP streaming. First of all, the limit of the existing TS in terms of flexible adaptation of multi-layer video is investigated, and the signaling by extending TS header is proposed to provide efficient adaptation in a TS level. The proposed extension utilizes the private data field in the adaptation field of TS header to signal scalability and/or view information, which enable us to support diverse adaptation that suits underlying constraints of client capabilities, network conditions and user preferences. In short, the extension enables adaptation of scalable video with full scalability as well as view selection of multiview video in a TS level while keeping backward compatibility with the existing TS syntax/semantics. The performance of the proposed extension is compared with the existing adaptation using PID (packet ID) in terms of efficiency and complexity of adaptation. Furthermore, the increase of TS overhead caused by proposed extension is analyzed and an extension scheme to minimized the overhead is proposed.