Implementing Audio-Over-IP from an IT Manager's Perspective
Total Page:16
File Type:pdf, Size:1020Kb
Load more
Recommended publications
-
On Ttethernet for Integrated Fault-Tolerant Spacecraft Networks
On TTEthernet for Integrated Fault-Tolerant Spacecraft Networks Andrew Loveless∗ NASA Johnson Space Center, Houston, TX, 77058 There has recently been a push for adopting integrated modular avionics (IMA) princi- ples in designing spacecraft architectures. This consolidation of multiple vehicle functions to shared computing platforms can significantly reduce spacecraft cost, weight, and de- sign complexity. Ethernet technology is attractive for inclusion in more integrated avionic systems due to its high speed, flexibility, and the availability of inexpensive commercial off-the-shelf (COTS) components. Furthermore, Ethernet can be augmented with a variety of quality of service (QoS) enhancements that enable its use for transmitting critical data. TTEthernet introduces a decentralized clock synchronization paradigm enabling the use of time-triggered Ethernet messaging appropriate for hard real-time applications. TTEther- net can also provide two forms of event-driven communication, therefore accommodating the full spectrum of traffic criticality levels required in IMA architectures. This paper explores the application of TTEthernet technology to future IMA spacecraft architectures as part of the Avionics and Software (A&S) project chartered by NASA's Advanced Ex- ploration Systems (AES) program. Nomenclature A&S = Avionics and Software Project AA2 = Ascent Abort 2 AES = Advanced Exploration Systems Program ANTARES = Advanced NASA Technology Architecture for Exploration Studies API = Application Program Interface ARM = Asteroid Redirect Mission -
Study of the Time Triggered Ethernet Dataflow
Institutionen f¨ordatavetenskap Department of Computer and Information Science Final thesis Study of the Time Triggered Ethernet Dataflow by Niclas Rosenvik LIU-IDA/LITH-EX-G{15/011|SE 2015-07-08 Linköpings universitet Linköpings universitet SE-581 83 Linköping, Sweden 581 83 Linköping Link¨opingsuniversitet Institutionen f¨ordatavetenskap Final thesis Study of the Time Triggered Ethernet Dataflow by Niclas Rosenvik LIU-IDA/LITH-EX-G{15/011|SE 2015-07-08 Supervisor: Unmesh Bordoloi Examiner: Petru Eles Abstract In recent years Ethernet has caught the attention of the real-time commu- nity. The main reason for this is that it has a high data troughput, 10Mbit/s and higher, and good EMI characteristics. As a protocol that might be used in real-time environments such as control systems for cars etc, it seems to fulfil the requirements. TTEthernet is a TDMA extension to normal Eth- ernet, designed to meet the hard deadlines required by real-time networks. This thesis describes how TTEthernet handles frames and the mathemat- ical formulas to calculate shuffle delay of frames in such a network. Open problems related to TTEthernet are also discussed. iii Contents 1 Introduction 1 2 Ethernet 2 2.1 Switching . 2 2.2 Ethernet frame format . 3 2.3 The need for TTEthernet . 4 3 TTEthernet 6 3.1 TTEthernet spec . 6 3.1.1 Protocol control frames . 6 3.1.2 Time-triggered frames . 7 3.1.3 Rate constrained frames . 9 3.1.4 Best effort frames . 10 3.2 Integration Algorithms . 10 3.2.1 Shuffling . 10 3.2.2 Preemption . -
Generating Synthetic Voip Traffic for Analyzing Redundant Openbsd
UNIVERSITY OF OSLO Department of Informatics Generating Synthetic VoIP Traffic for Analyzing Redundant OpenBSD-Firewalls Master Thesis Maurice David Woernhard May 23, 2006 Generating Synthetic VoIP Traffic for Analyzing Redundant OpenBSD-Firewalls Maurice David Woernhard May 23, 2006 Abstract Voice over IP, short VoIP, is among the fastest growing broadband technologies in the private and commercial sector. Compared to the Plain Old Telephone System (POTS), Internet telephony has reduced availability, measured in uptime guarantees per a given time period. This thesis makes a contribution towards proper quantitative statements about network availability when using two redun- dant, state synchronized computers, acting as firewalls between the Internet (WAN) and the local area network (LAN). First, methods for generating adequate VoIP traffic volumes for loading a Gigabit Ethernet link are examined, with the goal of using a minimal set of hardware, namely one regular desktop computer. pktgen, the Linux kernel UDP packet generator, was chosen for generating synthetic/artificial traffic, reflecting the common VoIP packet characteristics packet size, changing sender and receiver address, as well as typical UDP-port usage. pktgen’s three main parameters influencing the generation rate are fixed inter-packet delay, packet size and total packet count. It was sought to relate these to more user-friendly val- ues of amount of simultaneous calls, voice codec employed and call duration. The proposed method fails to model VoIP traffic accurately, mostly due to the cur- rently unstable nature of pktgen. However, it is suited for generating enough packets for testing the firewalls. Second, the traffic forwarding limit and failover behavior of the redun- dant, state-synchronized firewalls was examined. -
Comrex's Future Takes Shape with IP
www.tvtechnology.com/04-08-15 TV TECHNOLOGY April 8, 2015 21 TK Comrex’s Future Takes Shape With IP Company’s NAB booth to feature updated LiveShot, BRIC-Link II BY SUSAN ASHWORTH productions for television,” he said. “Over the last several decades we’ve been in LAS VEGAS—Building on its experience the radio space but we’ve had so many of in remote broadcasting technology, our radio customers move on to TV [and Comrex will come to the 2015 NAB Show adopted] our audio-over-IP codec, so we with IP on its mind, showcasing technology had a lot of requests to make something that the company sees as the future of live as portable and compact as our audio video broadcasting. products but for television.” Comrex will introduce the newest What’s especially compelling about this version of LiveShot, a system that allows segment of the market, he said, is that there broadcasters to tackle remote broadcast are people “who are doing really creative setups that would be tough with and unique broadcasts with our products traditional wired configurations. Using because [they] offer two-way video and Comrex ACCESS audio IP codecs, LiveShot return video to the field and intercom in a sends live HD video and audio over IP. The small little package.” system addresses technical inconsistencies For example, at a recent air show in in public Internet locales and provides Wisconsin, two wireless Comrex devices access to low-latency broadcast-quality were used by a local broadcaster for their live video streaming, including 3G, 4G, and multicamera fieldwork. -
IP Audio Coding with Introduction to BRIC Technology 2 Introduction by Tom Hartnett—Comrex Tech Director
IP Audio Coding With Introduction to BRIC Technology 2 Introduction By Tom Hartnett—Comrex Tech Director In 1992, when ISDN was just becoming available in the US, Comrex published a Switched 56/ISDN Primer which became, we were told, a very valuable resource to the radio engineer struggling to understand these new concepts. As POTS codecs and GSM codecs became viable tools, Comrex published similar primers. Now, with the gradual sunsetting of ISDN availability (and the migration of phone networks to IP based services), it not only makes sense for us to introduce a product based on Internet audio transfer, but again to publish all the relevant concepts for the uninitiated. The ACCESS product is the result of years of our research into the state of IP networks and audio coding algo- rithms. This has all been in ISDN is not a long term solution. the quest to do what we do The telephone network is changing. best, which is to leverage existing, available services Transition to IP is inevitable. to the benefit of our core Resistance is futile. customers—radio remote broadcasters. The heart of this product is called BRIC (Broadcast Reliable Internet Codec). While others have introduced hardware coined “IP Codecs,” this is the first product introduced that dares to use the wordInternet with a capital I. Given the challenges the public Internet presents, it’s no small boast to say that this product will perform over the majority of available connections. BRIC represents a change that is both desirable and inevitable for remotes. This change is inevitable because, as available connections move from old fashioned circuit switched to newer packet switched style, technology like ISDN and POTS codecs will begin to work less and less often. -
Developments in Audio Networking Protocols By: Mel Lambert
TECHNICAL FOCUS: SOUND Copyright Lighting&Sound America November 2014 http://www.lightingandsoundamerica.com/LSA.html Developments in Audio Networking Protocols By: Mel Lambert It’s an enviable dream: the ability to prominent of these current offerings, ular protocol and the basis for connect any piece of audio equip- with an emphasis on their applicability Internet-based systems: IP, the ment to other system components within live sound environments. Internet protocol, handles the and seamlessly transfer digital materi- exchange of data between routers al in real time from one device to OSI layer-based model for using unique IP addresses that can another using the long-predicted con- AV networks hence select paths for network traffic; vergence between AV and IT. And To understand how AV networks while TCP ensures that the data is with recent developments in open work, it is worth briefly reviewing the transmitted reliably and without industry standards and plug-and-play OSI layer-based model, which divides errors. Popular Ethernet-based proto- operability available from several well- protocols into a number of smaller cols are covered by a series of IEEE advanced proprietary systems, that elements that accomplish a specific 802.3 standards running at a variety dream is fast becoming a reality. sub-task, and interact with one of data-transfer speeds and media, Beyond relaying digital-format signals another in specific, carefully defined including familiar CAT-5/6 copper and via conventional AES/EBU two-chan- ways. Layering allows the parts of a fiber-optic cables. nel and MADI-format multichannel protocol to be designed and tested All AV networking involves two pri- connections—which requires dedicat- more easily, simplifying each design mary roles: control, including configur- ed, wired links—system operators are stage. -
The Essential Guide to Audio Over IP for Broadcasters 2 5
The Essential Guide To Audio OverIP FOR BROADCASTERS Powerful Performance | Powerful Control | Powerful Savings i 1. Why IP for Broadcast Audio? Reasons to Migrate to Audio over IP ..........................................................................................................8 1. Flexibility ..................................................................................................................................................................................8 2. Cost ...........................................................................................................................................................................................8 3. Scalability ................................................................................................................................................................................9 4. Reliability (yes really!) .........................................................................................................................................................9 5. Availability ..............................................................................................................................................................................9 6. Control and Monitoring .....................................................................................................................................................9 7. Network Consolidation ......................................................................................................................................................9 -
E-IPA-HX Audio-Over-IP Interface Card Eclipse HX Matrix Systems
E-IPA-HX Audio-over-IP Interface Card Eclipse HX Matrix Systems Linking People Together E-IPA-HX Interface Card The E-IPA-HX AoIP Interface card provides multiple IP Key Features and Benefits connection types for Eclipse® HX Matrix Intercom Systems, • Available in 16, 32, 48, and 64 port including support for AES67 and SMPTE ST2110 audio. cards (16 port upgrades) • Software definable port configuration Description to support multiple IP connection and The E-IPA-HX is a high density IP interface card that supports up to 64 Clear-Com standards intercom devices. The interface card can be used with the Eclipse HX-Delta Lite, • Supports up to 64 IP ports (endpoint HX-Delta, HX-Median, and HX-Omega matrix frames. Eclipse HX Configuration connections) and additionally up to 64 Software (EHX™) configures each port for its intended application and provides a FreeSpeak IP Transceivers dedicated IP Manager screen to monitor connections and add users. The E-IPA-HX card also supports SMPTE ST2110 and AES67 connectivity. • Compatible with Eclipse HX-Delta Lite, -Delta, -Median and -Omega Connection to FreeSpeak II and FreeSpeak Edge Beltpacks frames The E-IPA-HX card connects to FreeSpeak II® Beltpacks (1.9 or 2.4) in E1 mode via • Supports connection to V-Series and fiber to the FSII-SPL for the FSII-TCVR-24 or FSII-TCVR-19-XX transceivers. This V-Series Iris panels, Agent-IC mobile will allow up to 50 beltpacks and 10 transceivers per card. The E-IPA-HX card also app, FreeSpeak II wireless beltpacks, connects FreeSpeak II or FreeSpeak Edge™ devices via AES67 protocol to Eclipse HX LQ Series devices and Station-IC frames. -
Low-Latency Audio Over IP on Embedded Systems
Master Thesis Low-Latency Audio over IP on Embedded Systems by Florian Meier Start date: 02. April 2013 End date: 02. October 2013 Supervisor: Dipl.-Ing. Marco Fink Supervising Professor: Prof. Dr.-Ing. habil. Udo Zölzer Second Examiner: Prof. Dr. rer. nat. Volker Turau Abstract Transmission of audio data over networks, such as the Internet, is a widespread technology. Up to now, the primary application is voice trans- mission (Voice over IP). Although modern Voice over IP systems are de- signed to reduce the audio latency, it is still too high for the bidirectional transmission of live music. The construction of a low-latency music envi- ronment would enable distributed musical performances and "Jamming over IP". This thesis takes a step in this direction by building an Audio over IP system as an embedded system. All components needed for a jam- ming session over the Internet are integrated in a handy box on the basis of a Raspberry Pi. This required the development of a Linux kernel driver. Furthermore, a software for low-latency audio transmission is build and evaluated, considering audio quality, data rate, and the reduced computa- tional power of the embedded system. iii iv Declaration by Candidate I, FLORIAN MEIER (student of Informatik-Ingenieurwesen at Hamburg University of Tech- nology, matriculation number 20836390), hereby declare that this thesis is my own work and effort and that it has not been submitted anywhere for any award. Where other sources of information have been used, they have been acknowledged. Hamburg, 02. October 2013 Florian Meier v vi TABLE OF CONTENTS vii Table of Contents List of Figures ix List of Tables and Sourcecodes xi List of Symbols xiii 1 Introduction 1 2 State of the Art3 2.1 Effect of Latency for Musical Interaction . -
Overview on IP Audio Networking Andreas Hildebrand, RAVENNA Evangelist ALC Networx Gmbh, Munich Topics
Overview on IP Audio Networking Andreas Hildebrand, RAVENNA Evangelist ALC NetworX GmbH, Munich Topics: • Audio networking vs. OSI Layers • Overview on IP audio solutions • AES67 & RAVENNA • Real-world application examples • Brief introduction to SMPTE ST2110 • NMOS • Control protocols Overview on IP Audio Networking - A. Hildebrand # 1 Layer 2 Layer 1 AVB EtherSound Layer 3 Audio over IP Audio over Ethernet ACIP TCP unicast RAVENNA AES67 multicast RTP UDP X192 Media streaming Dante CobraNet Livewire Overview on IP Audio Networking - A. Hildebrand # 3 Layer 2 Layer 1 AVB Terminology oftenEtherSound Layer 3 Audio over IP • ambiguousAudio over Ethernet ACIP TCP unicast • usedRAVENNA in wrongAES67 context multicast RTP • marketingUDP -driven X192 Media streaming • creates confusion Dante CobraNet Livewire Overview on IP Audio Networking - A. Hildebrand # 4 Layer 2 Layer 1 AVB Terminology oftenEtherSound Layer 3 Audio over IP • ambiguousAudio over Ethernet ACIP TCP Audio over IP unicast • usedRAVENNA in wrongAES67 context multicast RTP • marketingUDP -driven X192 Media streaming • creates confusion Dante CobraNet Livewire Overview on IP Audio Networking - A. Hildebrand # 5 Layer 7 Application Application Application and Layer 6 Presentation protocol-based layers Presentation HTTP, FTP, SMNP, Layer 5 Session Session POP3, Telnet, TCP, Layer 4 Transport UDP, RTP Transport Layer 3 Network Internet Protocol (IP) Network Layer 2 Data Link Ethernet, PPP… Data Link Layer 1 Physical 10011101 Physical Overview on IP Audio Networking - A. Hildebrand # 10 Physical transmission Classification by OSI network layer: Layer 1 Systems Transmit Receive Layer 1 Physical 10011101 Physical Overview on IP Audio Networking - A. Hildebrand # 12 Physical transmission Layer 1 systems: • Examples: SuperMac (AES50), A-Net Pro16/64 (Aviom), Rocknet 300 (Riedel), Optocore (Optocore), MediorNet (Riedel) • Fully proprietary systems • Make use of layer 1 physical transport (e.g. -
Model 5421 Dante® Intercom Audio Engine
Model 5421 Dante® Intercom Audio Engine Key Features • 16-channel audio engine creates multiple • DDM support and AES67 compliant virtual party-line (PL) intercom circuits • PoE powered, Gigabit Ethernet interface • Dante audio-over-Ethernet technology • Configured using STcontroller application • Auto Mix for enhanced audio performance • Table-top, portable, or optional rack-mount • Supports Studio Technologies’ intercom installation beltpacks Overview The Model 5421 Dante® Intercom Audio Engine is a high- enclosure can be used stand-alone or mounted in one space performance, cost-effective, flexible solution for creating (1U) of a standard 19-inch rack enclosure with an optional party-line (PL) intercom circuits. It’s directly compatible rack-mount installation kit. To meet the latest interoper- with the Studio Technologies’ range of 1-, 2-, and 4-chan- ability standard the Model 5421’s Dante implementation nel Dante-enabled beltpacks and other interface-related meets the requirements of AES67 as well as supporting the products. The unit is suitable for use in fixed and mobile Dante Domain Manager (DDM) application. Using DDM, broadcast facilities, post-production studios, commercial compliance with ST 2110-30 may be possible. and educational theater environments, and entertainment The Model 5421 provides one 16-channel audio engine applications. which can be configured to provide from one to four “virtual” Only a Gigabit Ethernet network connection with Power- intercom circuits. The term “audio engine” was selected over-Ethernet (PoE) support is required for the Model to describe a set of audio input, processing, routing, and 5421 to provide a powerful resource in a variety of Dante output resources that can be configured to support spe- applications. -
Hasseb Audio Over Ethernet XLR Instructions Manual Version
September HASSEB AUDIO OVER ETHERNET XLR INSTRUCTIONS MANUAL 19, 2018 VERSION 1.0 HASSEB AUDIO OVER ETHERNET XLR hasseb Audio over Ethernet XLR is an easy to use and portable device used to send lossless, realtime microphone audio signal using Ethenet network. The device is compatible with AES67 standard and can be used as a standalone device or together with other AES67 compatible devices. Two 3-pin XLR audio connectors are used to connect microphones to Ethernet netowork. The device has two input channels with a high quality analog amplifier and A/D converter to digitize the analog microphone signals. In addition, the device has a 48 V phantom source for microphones requiring phantom power. The device is configured using a web user interface and mDNS (multicast Domain Name System) protocol is supported to easily find the device IP addresses from the network. For professional grade network audio systems PTP (Precision Time Protocol) based time synchronization is required. The device can act as IEEE1588 grand master to provide synchronization clock signal to the network. INSTALLATION The device can be powered through the Ethernet cable using a Power over Ethernet capable switch of through a micro USB connector with an external 5 V USB power supply. DHCP (Dynamic Host Configuration Protocol) support is enabled by default, so the device will assign an IP address automatically. All hasseb Audio over Ethernet XLR devices and most other AES67 devices and their names and IP addresses connected to the network can be found using any software, capable of searching the network for mDNS supported devices.