Duplicating RTP streams

Packet loss is undesirable for real-time multimedia sessions but can occur due to a variety of reasons including unplanned network outages. In unicast transmissions, recovering from such an outage can be difficult depending on the outage duration, due to the potentially large number of missing packets. In multicast transmissions, recovery is even more challenging as many receivers could be impacted by the outage. For this challenge, one solution that does not incur unbounded delay is to duplicate the packets and send them in separate redundant streams, provided that the underlying network satisfies certain requirements. This document explains how Real-time Transport Protocol (RTP) streams can be duplicated without breaking RTP or RTP Control Protocol (RTCP) rule

Synchronization of streamed audio between multiple playback devices over an unmanaged IP network

When designing and implementing a prototype supporting inter-destination media synchronization – synchronized playback between multiple devices receiving the same stream – there are a lot of aspects that need to be considered, especially when working with unmanaged networks. Not only is a proper streaming protocol essential, but also a way to obtain and maintain the synchronization of the clocks of the devices. The thesis had a few constraints, namely that the server producing the stream should be written for the .NET-platform and that the clients receiving it should be using the media framework GStreamer. This framework provides methods for both achieving synchronization as well as resynchronization. As the provided resynchro- nization methods introduced distortions in the audio, an alternative method was implemented. This method focused on minimizing the distortions, thus maintain- ing a smooth playback. After the prototype had been implemented, it was tested to see how well it performed under the influence of packet loss and delay. The accuracy of the synchronization was also tested under optimal conditions using two different time synchronization protocols. What could be concluded from this was that a good synchronization could be maintained on unloaded networks using the proposed method, but when introducing delay the prototype struggled more. This was mainly due to the usage of the Network Time Protocol (NTP), which is known to perform badly on networks with asymmetric paths.When working with synchronized playback it is not enough just obtain- ing it – it also needs to be maintained. Implementing a prototype thus involves many parts ranging from choosing a proper streaming protocol, to handling glitch free resynchronization of audio. Synchronization between multiple speakers has a wide area of application, ranging from home entertainment solutions to big malls where announcements should appear synchronized over the entire perimeter. In order to achieve this, two main parts are involved: the streaming of the audio, and the actual synchronization. The streaming itself poses problems mostly since the prototype should not only work on dedicated networks, but rather on all kinds, such as the Internet. As the information over these networks are transmitted in packets, and the path from source to destination crosses many sub networks, the packets may be delayed or even lost. This may create an audible distortion in the playback. The next part is the synchronization. This is most easily achieved by putting a time on each packet stating when in the future it should be played out. If then all receivers play it back at the specified time, synchronization is achieved. This however requires that all the receivers share the idea of when a specific time is – the clocks at all the receivers must be synchronized. By using existing software and hardware solutions, such as the Network Time Protocol (NTP) or the Precision Time Protocol (PTP), this can be accomplished. The accuracy of the synchronization is therefore partly dependent on how well these solutions work. Another valid aspect is how accurate the synchronization must be for the sound to be perceived as synchronized by humans. This is usually in the range of a few tens of milliseconds to five milliseconds depending on the sound. When a global time has been distributed to all receivers, matters get more complicated as there is more than one clock to consider at each receiver. Apart from the previously mentioned clock, now called the ’system clock’, there is also an audio clock, which is a hardware clock positioned on the sound card. This audio clock decides the rate at which media is played out. Altering the system clock to synchronize it to a common time is one thing, but altering the audio clock while media is being played will inevitably mean a jump in the playback, and thus a distortion. Although an initial synchronization can be achieved, the two clocks will over time tick in slightly different pace, thus drifting away from each other. This creates a need for the audio clock to continuously correct itself to follow the system clock. In the media framework GStreamer, used for handling the media at the re- ceivers, two alternatives to solve the correction problem were available. Quick evaluations of these two methods however showed that either audible glitches or ’oscillations’ occurred in the sound, when the clocks were corrected. A new method, which basically combines the two existing, was therefore implemented. With this method the audio clock is continuously corrected, but in a smaller and less aggressive way. Listening tests revealed much smaller, often not audible, distortions, while the synchronization performance was at par with the existing methods. More thorough testing showed that the synchronization over networks with light traffic was in the microsecond-range, thus far below the threshold of what will appear as synchronized. During worse conditions – simulated hostile environments – the synchronization quickly reached unacceptable levels though. This was due to the previously mentioned NTP, and not the implemented method on the other hand

Prevalent Network Threats and Telecommunication Security Challenges and Countermeasures in VoIP Networks

Due to the recent global popularity gained by VoIP network while many organisations/industries are employing it for their voice communication needs, optimal security assurance has to be provided to guarantee security of their data/information against present day teeming security threats and attacks prevalent in IP-based networks. This research paper has critically investigated and analysed most of the security challenges associated with VoIP systems and traditional IP data networks; and has proposed several defence measures which if designed and implemented will prevent most (if not all) of the security threats plaguing these networks. Keywords: Network security, VoIP, Computer attack, Security threats, SIP, H.323, Defence measures, IPSec

Control Synchronous Web-Based Training Using Web Services

With the rapidly advancing technologies, training has been vital to keep companies competitive. Web-based training grows rapidly and attracts more attention for its most flexible manner. Virtual classroom is a form of synchronous web-based training. It provides real-time interactivity in learning process. I have developed a virtual classroom that uses Web services to control the audio/video transmission, chat box, whiteboard, and synchronous HTML presentation. Compared to an early implementation of the virtual classroom based on the Jini network, my Web-service based implementation has a significantly different control structure. My implementation has better interoperability

SVCEval-RA: an evaluation framework for adaptive scalable video streaming

[EN] Multimedia content adaption strategies are becoming increasingly important for effective video streaming over the actual heterogeneous networks. Thus, evaluation frameworks for adaptive video play an important role in the designing and deploying process of adaptive multimedia streaming systems. This paper describes a novel simulation framework for rate-adaptive video transmission using the Scalable Video Coding standard (H.264/SVC). Our approach uses feedback information about the available bandwidth to allow the video source to select the most suitable combination of SVC layers for the transmission of a video sequence. The proposed solution has been integrated into the network simulator NS-2 in order to support realistic network simulations. To demonstrate the usefulness of the proposed solution we perform a simulation study where a video sequence was transmitted over a three network scenarios. The experimental results show that the Adaptive SVC scheme implemented in our framework provides an efficient alternative that helps to avoid an increase in the network congestion in resource-constrained networks. Improvements in video quality, in terms of PSNR (Peak Signal to Noise Ratio) and SSIM (Structural Similarity Index) are also obtained.Castellanos Hernández, WE.; Guerri Cebollada, JC.; Arce Vila, P. (2017). SVCEval-RA: an evaluation framework for adaptive scalable video streaming. Multimedia Tools and Applications. 76(1):437-461. doi:10.1007/s11042-015-3046-yS437461761Akhshabi S, Begen AC, Dovrolis C (2011) An experimental evaluation of rate-adaptation algorithms in adaptive streaming over HTTP. In: Proceedings of the second annual ACM conference on Multimedia systems. ACM, pp 157–168Alabdulkarim MN, Rikli N-E (2012) QoS Provisioning for H.264/SVC Streams over Ad-Hoc ZigBee Networks Using Cross-Layer Design. In: 8th International Conference on Wireless Communications, Networking and Mobile Computing (WiCOM). pp 1–8Birkos K, Tselios C, Dagiuklas T, Kotsopoulos S (2013) Peer selection and scheduling of H. 264 SVC video over wireless networks. In: Wireless Communications and Networking Conference (WCNC), 2013 IEEE. pp 1633–1638Castellanos W (2014) SVCEval-RA - An Evaluation Framework for Adaptive Scalable Video Streaming. In: SourceForge Project. http://sourceforge.net/projects/svceval-ra/ . Accessed 1 May 2015Castellanos W, Guerri JC, Arce P (2015) A QoS-aware routing protocol with adaptive feedback scheme for video streaming for mobile networks. Comput Commun. http://dx.doi.org/10.1016/j.comcom.2015.08.012Castellanos W, Arce P, Acelas P, Guerri JC (2012) Route Recovery Algorithm for QoS-Aware Routing in MANETs. Springer Berlin Heidelberg, Bilbao, pp. 81–93Chikkerur S, Sundaram V, Reisslein M, Karam LJ (2011) Objective video quality assessment methods: A classification, review, and performance comparison. Broadcast, IEEE Trans on 57:165–182Choupani R, Wong S, Tolun M (2014) Multiple description coding for SNR scalable video transmission over unreliable networks. Multimed Tools Appl 69:843–858. doi: 10.1007/s11042-012-1150-9CISCO Corp. (2014) Cisco Visual Networking Index Forecast and Methodology. In: White Paper. http://www.cisco.com/c/en/us/solutions/collateral/service-provider/ip-ngn-ip-next-generation-network/white_paper_c11-481360.pdf.Dai M, Zhang Y, Loguinov D (2009) A unified traffic model for MPEG-4 and H. 264 video traces. IEEE Trans Multimedia 11:1010–1023Detti A, Bianchi G, Pisa C, et al. (2009) SVEF: an open-source experimental evaluation framework for H.264 scalable video streaming. 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KSII Trans Internet Inf Syst (TIIS) 6:377–392. doi: 10.3837/tiis.2012.01.021Ke C-H, Shieh C-K, Hwang W-S, Ziviani A (2008) An Evaluation Framework for More Realistic Simulations of MPEG Video Transmission. J Inf Sci Eng 24:425–440Klaue J, Rathke B, Wolisz A (2003) Evalvid–A framework for video transmission and quality evaluation. In: Computer Performance Evaluation. Modelling Techniques and Tools. Springer, pp 255–272Le TA, Nguyen H (2014) End-to-end transmission of scalable video contents: performance evaluation over EvalSVC—a new open-source evaluation platform. Multimed Tools Appl 72:1239–1256. doi: 10.1007/s11042-013-1444-6Lie A, Klaue J (2008) Evalvid-RA: trace driven simulation of rate adaptive MPEG-4 VBR video. Multimedia Systems 14:33–50. doi: 10.1007/s00530-007-0110-0Moving Pictures Experts Group and ITU-T Video Coding Experts Group (2011) H. 264/SVC reference software (JSVM 9.19.14) and Manual.Nightingale J, Wang Q, Grecos C (2014) Empirical evaluation of H.264/SVC streaming in resource-constrained multihomed mobile networks. Multimed Tools Appl 70:2011–2035. doi: 10.1007/s11042-012-1219-5Parmar H, Thornburgh M (2012) Real-Time Messaging Protocol (RTMP) Specification. AdobePolitis I, Dounis L, Dagiuklas T (2012) H. 264/SVC vs. H. 264/AVC video quality comparison under QoE-driven seamless handoff. Signal Process Image Commun 27:814–826Pozueco L, Pañeda XG, García R, et al. (2013) Adaptable system based on Scalable Video Coding for high-quality video service. Comput Electr Eng 39:775–789. doi: 10.1016/j.compeleceng.2013.01.015Pozueco L, Pañeda XG, García R, et al. (2014) Adaptation engine for a streaming service based on MPEG-DASH. 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DiversiFi: Robust Multi-Link Interactive Streaming

ABSTRACT Real-time, interactive streaming for applications such as audio-video conferencing (e.g., Skype) and cloud-based gaming depends critically on the network providing low latency, jitter, and packet loss, much more so than on-demand streaming (e.g., YouTube) does. However, WiFi networks pose a challenge; our analysis of data from a large VoIP provider and from our own measurements shows that the WiFi access link is a significant cause of poor streaming experience. To improve streaming quality over WiFi, we present DiversiFi, which takes advantage of the diversity of WiFi links available in the vicinity, even when the individual links are poor. Leveraging such cross-link spatial and channel diversity outperforms both traditional link selection and the temporal diversity arising from retransmissions on the same link. It also provides significant gains over and above the PHY-layer spatial diversity provided by MIMO. Our experimental evaluation shows that, for a client with two NICs, enabling replication across two WiFi links helps cut down the poor call rate (PCR) for VoIP by 2.24x. Finally, we present the design and implementation of DiversiFi, which enables it to operate with single-NIC clients, and with either minimally modified APs or unmodified APs augmented with a middlebox. Over 61 runs, where the baseline average PCR is 4.9%, DiversiFi running with a single NIC, switching between two links, helps cut the PCR down to 0%, while duplicating wastefully only 0.62% of the packets and impacting competing TCP throughput by only 2.5%. Thus, DiversiFi provides the benefit of multi-link diversity for real-time interactive streaming in a manner that is deployable and imposes little overhead, thereby ensuring coexistence with other applications

West Tennessee River Basin Authority (WTRBA) Annual Report 2020

https://digitalcommons.memphis.edu/govpubs-tn-dept-environment-conservation-wtrba-annual-report/1002/thumbnail.jp