Michael Maruschke

Michael Maruschke
Hochschule für Telekommunikation Leipzig | HfTL · Computer Science

Diplom-Ing (FH)

About

31
Publications
18,757
Reads
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144
Citations
Citations since 2016
17 Research Items
119 Citations
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Additional affiliations
February 2011 - present
Hochschule für Telekommunikation Leipzig
Position
  • Lecturer
Description
  • Network Infrastructures and Protocols
January 2009 - present
Hochschule für Telekommunikation Leipzig
Position
  • Researcher

Publications

Publications (31)
Conference Paper
Full-text available
Recent audio codecs enable high quality signals up to fullband (20 kHz) which is usually associated with the maximal audible bandwidth. Following previous studies on speech coding assessment, we survey in this novel study the music coding ability of two real-time codecs with fullband capability – the IETF standardized Opus codec as well as the 3GPP...
Conference Paper
Full-text available
In this paper, the authors propose an extensible (Open) Graphical User (Flow) Interface called OpenGUFI. Open- GUFI is a Software-Defined Networking application that provides a visual abstraction of the underlying network. The Visualizer has been implemented and tested in an SDN-enabled Wireless Testbed based on the Odin framework. As a use case a...
Conference Paper
This paper describes a method to support QoS using WebRTC-based Clients integrated in mobile devices conformed to 3GPP Rel. 8 Public Land Mobile Network infrastructure. To enrich the experience for mobile customers using this upcoming technology, the article focuses on the merging of the technical capabilities arising from both the Evolved Packet S...
Conference Paper
Full-text available
This paper reviews the recent activities of Open Source Software (OSS) adoptions by governments, education sectors, and businesses in Germany, Brazil, and India. It looks at their motivation and focuses on the selected developments of OSS. Typical applications will be discussed as well as consequences from using OSS.
Conference Paper
Full-text available
This article presents the results of WebRTC trials conducted in a partnership between Telekom Innovation Laboratories (T-Labs) and PT Inovação e Sistemas under the project called WONDER. The main achievement of WONDER project was the design and validation of the new "Signalling-On-the-fly" (SigOfly) concept. This concept ensures interoperability be...
Conference Paper
Full-text available
Common applications of an unmanned aerial vehicle (UAV, aerial drone) utilize the capabilities of mobile image or video capturing, whereas our article deals with acoustic-related scenarios. Especially for surveillance tasks, e.g. in disaster management or measurement of artificial environmental noise in large industrial areas, an UAV-based acoustic...
Patent
Full-text available
The present invention relates to support Quality of Service (QoS) using Web Real Time Communication (WebRTC) based clients integrated in mobile devices conformed to 3GPP Release 8 Public Land Mobile Network (PLMN) infrastructure.
Chapter
Full-text available
Recently, the use of Web Real-Time Communication (WebRTC) technology in communication applications has been increasing significantly. The users of IP-based telephony require excellent audio quality. However, in WebRTC-based audio calls the audio assessment is challenging due to the specific functioning principles of WebRTC, such as security requireme...
Conference Paper
Full-text available
The main question that this paper intends to answer is what telecommunication operators could do for their mobile network evolution until 5 th Generation (5G) mobile networks are deployed for mass-market and beyond 2020. Especially in scope is policy-related functionality in the core network. The roadmap for mobile networks should take into account...
Conference Paper
Full-text available
We investigated 14 German politicians to systematically analyze typical boundary conditions in the perceptive charisma assessment, such as age and gender of speakers and evaluators. Beyond, we surveyed the correlation between the perceptive results and some prosodic correlates. The test data included 84 speech samples of 7 female and 7 male speaker...
Conference Paper
Full-text available
Speech and audio codecs are implemented in a variety of multimedia applications, and multichannel sound is offered by first streaming or cloud-based services. Beside the objective of perceptual quality, coding-related research is focused on low bitrate and minimal latency. The IETF-standardized Opus codec provides a high perceptual quality, low lat...
Conference Paper
Full-text available
Mobile-Edge Computing (MEC) is a highly relevant topic and is continuing in the standardisation process of a dedicated ETSI Industry Specification Group. In addition to their service scenarios, we present new emerging use cases in the context of MEC. We introduce and categorise approaches how to move a service function between hosting environments....
Conference Paper
Full-text available
The article describes the first perceptual quality study of 5.1 surround sound that has been processed by the Opus codec standardised by the Internet Engineering Task Force (IETF). All listening sessions with up to five subjects took place in a slightly sound absorbing laboratory – simulating living room conditions. For the assessment we conducted...
Conference Paper
Full-text available
Enabling a natural (human-like) spoken conversation with technical systems requires affective information, contained in spoken language, to be intelligibly transmitted. This study investigates the role of speech and music codecs for affect intelligibility. A decoding and encoding of affective speech was employed from the well-known EMO-DB corpus. U...
Conference Paper
Full-text available
This article deals with the representation of prosodic attributes in coded speech which is less-studied. Common models in speech coding assume that there is no relevant influence of prosodic variation on perceived quality and content of coded speech under suitable operating conditions. Our experiments included a listening test and the instrumental...
Conference Paper
Full-text available
In human-computer interaction, affective measures like emotion assessment are an important means to steer and improve the dialog flow, which will be an integral part of automatic interaction management. In this context, we evaluate the intelligibility of affective speech both by inter-rater reliability (IRR) in human voting and by the ITU- recommen...
Patent
Full-text available
The object of the present invention is to provide a method and a system for integrating WebRTC based clients into IMS alternative to standardized solution and with a reduced traffic load produced by SIP Register signalling messages.
Conference Paper
Full-text available
Highly-efficient technologies or algorithms, including speech & audio coding, play an important role in human-machine interaction, human communication and user interface design. This contribution summarizes selected results from our performance studies on speech and audio codecs – mainly within the internet-oriented WebRTC scenario in comparison to...
Conference Paper
Full-text available
Zur Erreichung der Interoperabilität von Endnutzern, welche unterschiedliche Audiocodecs nutzen, wird eine Umwandlung (Transcoding) zwischen den jeweils zum Einsatz kommenden Codecs durchgeführt. Dieser Beitrag beschäftigt sich mit der Frage, welchen Einfluss die durch das Transcoding entstehende Verzögerung auf das Ende-zu-Ende-Delay der Kommunika...
Conference Paper
Full-text available
The Opus codec is used in several applications fields of speech and audio communication. This article describes the instrumental quality assessment of Opus-coded speech in a web browser-based real-time communication using POLQA and AQuA method. Furthermore, we tested Opus with mixed vocal and music signals and also performed a perceptual test. WebR...
Conference Paper
Full-text available
In this paper, the authors propose an extensible (Open) Graphical User (Flow) Interface called OpenGUFI. Open- GUFI is a Software-Defined Networking application that provides a visual abstraction of the underlying network. The Visualizer has been implemented and tested in an SDN-enabled Wireless Testbed based on the Odin framework. As a use case a...
Conference Paper
Full-text available
The Internet Engineering Task Force (IETF) – the open Internet standards-development body – considers the Opus codec as a highly versatile audio codec for interactive voice and music transmission. In this review we survey the dynamic functioning of the Opus codec within a Web Real-Time Communication (WebRTC) framework based on the Google Chrome bro...
Article
Full-text available
This paper describes the considerable role of Quality of Service (QoS) for Web Real-Time Communication (WebRTC) clients connected with an IP Multimedia Subsystem (IMS)-based IP Television (IPTV) infrastructure. To raise the quality of experience for IPTV customers, the article focuses on the merging of the technical capabilities arising from both t...
Conference Paper
Full-text available
This paper describes the potential to combine IP Multimedia Subsystem (IMS)-based IP Television (IPTV) services with a future-oriented web browser technology, the Web Real-Time Communication (WebRTC). To enrich the quality of experience for residential customers using this upcoming technology, the article focuses on the merging of the technical cap...
Conference Paper
Full-text available
This paper describes an implementation for monitoring the QoS and expecting the QoE of a voice communication in a Real-time Transport Protocol (RTP) based telecommunication environment. The resulting QoS parameters are evaluated; the QoE is determined with the E-Model and processed for graphical presentation. With the use of some open-source progra...
Conference Paper
Full-text available
For upcoming decade, IP Multimedia Subsystem (IMS) is preferred communication technology focused on delivering multimedia services. With its service-oriented and layered architecture it enables provision of variety of services and adoption 3 rd party providers' services. Communication network operators are loosing market share of service providers...
Conference Paper
Full-text available
This paper describes an approach for combining Session Initiation Protocol (SIP) based voice communication with Extensible Messaging and Presence Protocol (XMPP) based presence enhancements. The actual role of SIP and XMPP in the Internet Protocol (IP) based communication was analyzed, especially from the telecommunication carrier (Telco) point of...
Conference Paper
Full-text available
This paper illustrates both the detailed theoretical setup of the Interconnection Border Gateway Function (IBGF) and its technical implementation using Open Source Software components. Moreover, a model is proposed to handle different traffic classes according to their corresponding Quality of Service (QoS) requirements while also forwarding “best...
Chapter
Full-text available
This paper describes an approach for interworking scenarios between Session Initiation Protocol (SIP) based and non SIP based frameworks (e.g. web services) in case of the presence management service. The characteristics of the concept of a centralized presence management will be introduced. Based on the Parlay X web service framework a presence ma...
Conference Paper
Full-text available
This paper describes the integration of the presence management service into existing IP Multimedia Subsystem (IMS) laboratory infrastructures. Both laboratories - NGNlab in Bratislava and the laboratory in Leipzig - use the Fraunhofer OpenIMS as core components for their testing IMS environment (i.e. Call Session Control Functions (CSCF), Home Sub...

Questions

Questions (3)
Question
In ITU-T Y.2012, Supplement 1, are outlined the the "functions related to signalling path"; amongst other a "DSP service control; Codec negotiation and control of lower layer service"
Does anyone kwo, what does it means DSP in this context: Digital signal processing or is it a spelling mistake for Session Description Protocol (SDP)??
Answers are helpful to understand this standardization phrase at the above.
Thanks
Michael
Question
Audio Transcoding Units have to process the conversion of the incoming audio codec type A towards the outgoing audio codec type B. We like to measure the delay time for this process.
Question
The EVS audio Codec is standardized by 3GPP for SuperWideband and Fullband Voice Services in 4G networks. 

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Cited By

Projects

Projects (3)
Archived project
Project
The projects deals with the question, how we can realize dynamic growthing and dynamic variable network infrastructure like Evolved Packet Systems (EPS), toward a 5G ready network. Mobile Edge Cloud Computing could be one component on this way.