Árpád Huszák

Analysing GOP structure and packet loss effects on error propagation in MPEG-4 video streams

Video streaming applications are commonly used in both wired and wireless environment; however, wireless links are burden by higher packet loss ratio and delay variation. In order to make video transmission possible in wireless networks MPEG video coding is usually used to reach the bandwidth constrains of the links. The video quality and compression ratio depends on Group of Pictures (GOP) structure, but it also affects the distortion sensitivity of the video stream due to packet losses. In this paper the correlation between GOP size, packet loss ratio and video quality is investigated. By increasing the distance between the reference frames the effectiveness of coding can be improved, but on the other hand the effect of error propagation due to packet losses also increases. Our aim was to find the optimal GOP structure to maximize the coding efficiency and minimize the quality distortion due to error propagation. We have implemented a simulation tool to make the analysis of differently structured video streams transmitted over lossy channels.

Source controlled semi-reliable multimedia streaming using selective retransmission in DCCP/IP networks

In the past few years we have witnessed an explosive growth in the usage of media streaming applications. The newly appeared audio/video applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. Retransmission-based error recovery is considered inappropriate for multimedia applications, because of its latency. This solution can be attractive because it requires minimal network bandwidth, processing cost and efficiently improves the stream quality. Despite its latency, retransmission can be used successfully in many cases, especially if playout buffering is employed. Only the successfully retransmitted packets will improve the quality parameters of the multimedia stream, therefore it is worth to examine which packets should be retransmitted. In this paper a source controlled selective retransmission algorithm is presented with a decision algorithm based on the actual RTT and sending rate determined by the TFRC. In our scheme the transmitter determines the playout delay caused by the playout buffer using the proposed Flood method. The needed information about the network congestion state and the network delay are provided by the TFRC (TCP Friendly Rate Control) algorithm. Our proposal does not need additional administration messages because the decision procedure and its inputs are at the transmitter. The obtained results show that significant quality improvement is possible with the proposed selective retransmission scheme.

Content-aware interface selection method for multi-path video streaming in best-effort networks

Todays mobile devices are equipped with multiple interfaces to make the connection possible to different type of networks. In order to efficiently utilize the interface capabilities, multi-path streaming can be used. Resource intensive applications can deliver high bitrate streams over multiple paths by cumulating the available bandwidth of the different subpaths. In this paper we propose a multi-path streaming method that chooses a set of paths maximizing the overall quality at the client. While the available paths have different bandwidth, delay and loss probability constrains, the packet distributor must take the video packet importance and the dependencies between packets into account. Transmitting the reference video frames on the most reliable links will decrease the loss probability of important data packets and increase the measured video quality. Both analytical and simulation analysis were performed to examine the behavior of the presented content-aware interface selection model. The results show that the proposed solution provides higher video quality than common scheduling algorithms.

Source Controlled and Delay Sensitive Selective Retransmission Scheme for Multimedia Streaming

Multimedia applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. The wireless network characteristics make the deployment of these applications more challenging. Retransmission-based error recovery is considered inappropriate for multimedia applications, because of its latency. However, this solution can be attractive because it requires minimal network bandwidth and processing cost as well as improvement of the video quality. Despite its latency, retransmission can be used successfully in many cases, especially if playout buffering is employed. In this paper we propose a source controlled and playout time oriented retransmission scheme for multimedia streaming. Due to source controlling the receiver do not need to send additional retransmission request messages. The other advantage of transmitter controlled decision algorithm is that all the needed information is available at the source due to DCCP transport protocol. Therefore no additional network measurements and probe messages are needed. The results show that our scheme reduces traffic overhead generated by retransmission and hardly degrades the number of unsuccessfully retransmitted packets at all.

On SCTP multihoming performance in native IPv6 UMTS-WLAN environments

Multihoming is one of the most attractive features of SCTP (Stream Control Transmission Protocol) aiming to make this prosperous transport scheme competitive in mobile environments when the mobile hosts are equipped with multiple interfaces. The complementary characteristic of the 3G UMTS (Universal Mobile Telecommunications System) and WLAN (Wireless Local Area Network) architectures motivates operators to integrate these two successful technologies. Recently spreading wireless devices are increasingly provided with multiple networking interfaces such enabling end-users to access Internet services using e.g. the WLANs higher bandwidth wherever available, and connecting to the UMTS network in other cases. This paper investigates the performance of multihomed SCTP hosts through extensive experimental studies in such an integrated heterogeneous environment. In order to do this we designed and implemented a real native IPv6 UMTS-LAN testbed equipped with novel IPv6-based mobile technologies, such providing ideal conditions for SCTP multihoming performance analysis. In this testbed architecture we analyzed numerous settings to measure the handover behavior of the protocol in terms of handover effectiveness, link changeover characteristics, throughput and transmission delay. As our results show, accurate SCTP parameter setup can significantly decrease the handover delay and eliminate the data transmission interrupts.

Novel results on SCTP multihoming performance in native IPv6 UMTS-WLAN environments

Multihoming is one of the most attractive features of stream control transmission protocol (SCTP) aiming to make this prosperous transport scheme competitive in mobile environments when the mobile hosts are equipped with multiple interfaces. The complementary characteristic of the 3G universal mobile telecommunications system (UMTS) and wireless local area network (WLAN) architectures motivates operators to integrate these two successful technologies. Recently spreading wireless devices are increasingly provided with multiple networking interfaces such enabling end-users to access internet services using e.g., the WLANs higher bandwidth wherever available, and connecting to the UMTS network in other cases. This paper investigates the performance of multihomed SCTP hosts through extensive experimental studies in such an integrated heterogeneous environment. In order to do this, we designed and implemented a real native IPv6 UMTS-WLAN testbed equipped with novel IPv6-based mobile technologies, such providing ideal conditions for SCTP multihoming performance analysis. In this testbed architecture we analysed numerous settings to measure the handover behaviour of the protocol in terms of handover effectiveness, link changeover characteristics, throughput and transmission delay. As our results show, accurate SCTP parameter setup can significantly decrease the handover delay and eliminate the data transmission interrupts.

QoS provision in IP based mobile networks

While IP is declared as the key technology of the future’s wired and mobile communication, the currently used version of IP, IPv4 itself is not suitable to be used in mobile scenarios. Next generation mobile users require special support to provide connectivity, although they change their place of attachment to the network frequently [1]. In our work, we have created a network design algorithm and an agent (GMA/MAP) router selection algorithm in Regional Registration and Hierarchical Mobile IPv6 to optimise the handover management in IP based next generation mobile networks. Our research is supported by ETIK (Inter-University Center for Telecomunications and Informatics).

Congestion Control and Adaptive Retransmission for Multimedia Streaming over Wireless Networks

In wireless networks, congestion control, alone, may not be enough to ensure good quality of multimedia streaming and efficient utilization of the network. Packet losses due to the high bit error rate not only degrade the multimedia quality, but render the current congestion control algorithms as inefficient: these algorithms back-off on every packet loss even when there is no congestion. In this paper we study the case of pre-recorded multimedia streaming over wireless networks. We integrate the congestion control schemes with an adaptive retransmission scheme in order to selectively retransmit some lost multimedia packets. Moreover, we integrate a wireless loss estimation scheme to improve the efficiency of congestion control. Our results show that the integrated scheme improves the multimedia quality due to the retransmission and recovery of some of the lost multimedia data. Moreover, the congestion control schemes, and hence the integrated retransmission scheme, show improved performance because of the wireless loss estimation.

DCCP-based multiple retransmission technique for multimedia streaming

Retransmission-based error recovery is the simplest technique to minimize the overall packet loss ratio in order to increase the quality of the applications. Multimedia applications are becoming increasingly popular in IP networks, while in mobile environment the limited bandwidth and the higher error rate arise in spite of its popularity. Retransmission can be also used for loss recovery in media applications, but the number of retransmissions is limited by the playout buffer and the recent network delay. In this paper we present a source controlled adaptive multiple retransmission scheme for multimedia streaming. Due to source controlling the receiver do not need to send additional retransmission request messages. The other advantage of transmitter controlled decision algorithm is that the necessary RTT information is available at the source due to DCCP transport protocol. Therefore no additional network measurements and probe messages are needed. With the knowledge of the actual network delay and the playout buffer delay, we are able to determine the number of retransmission attempts without late delivery of packets.

Predictive multicast group management for free viewpoint video streaming

Free viewpoint video (FVV) is a new approach of interactive streaming services, where users are able to freely change their viewpoint. The desired viewpoint is synthetized from two or more camera views that must be delivered to the users depending on their continuously changing perspective. Multicast delivery of camera streams is an appropriate solution, however due to network latency and frequent changes of the viewpoint, the required camera streams may arrive too late, interrupting the FVV synthesis and playout. In this paper a seamless FVV streaming scheme is presented based on user viewpoint prediction. In order to avoid the starving of the FVV synthesizer, we propose to use threshold areas to prefetch the camera views that will be probably required for the viewpoint generation. We have formulated the calculation steps of the threshold values in order to minimize the starvation ratio and its duration. The obtained simulation results show that using the predictive multicast management scheme, the clients receive the required camera views in time in more than 95% of the cases. Moreover, we showed that the number of used FVV cameras and the number of clients have significant impact on the performance of the FVV service.