Network loudspeaker

Last updated

A conventional loudspeaker is an electromechanical transducer that converts an electrical signal into sound. If locally powered, this can also be termed an active loudspeaker, meaning it contains an audio power amplifier that drives the loudspeaker. A network loudspeaker implies the ability to send audio to such a device from a network connection, usually over an Ethernet network or the Internet. In many cases this type of speaker also contains digital signal processing (DSP) to provide the audio crossover and other tonal functions that exist in conventional speakers. Network speakers are also known as IP speakers. [1] In many cases the IP speaker is created from an IP audio endpoint  — a device with the requisite network connection and ability to process audio packets, but without the actual physical speaker portion — that provides amplified audio to a conventional loudspeaker or unamplified audio (i.e. line-out) to an amplified speaker or system.

Contents

History

Network audio was first introduced in 1983 by John Detreville and W. David Sincoskie of Bell Labs in the IEEE paper "A Distributed Experimental Communications System". [2] Subsequently, in 1988, Polle T. Zellweger, Douglas B. Terry and Daniel C. Swinehart of Xerox PARC introduced audio over Ethernet at the 2nd IEEE Conference in a paper entitled "An Overview of the Etherphone System and its Applications". [3] The very closely related technology (nearly synonymous), Voice over IP (VoIP), which generally consists at a minimum of a microphone on one end, a speaker on the other end, with a network connecting them; began widespread use in 1998 with PCs, followed shortly after by dedicated hardware (phones with built-in VoIP capabilities). Subsequently, the first Squeezebox, using networked audio, was released in 2001 and Philips released its IP audio device also in 2001, the FW-i1000. Countless IP audio devices have since proliferated into most major audio markets.

Designs

IP Multicast Multicast.svg
IP Multicast
IP Unicast Unicast.svg
IP Unicast

The audio content played by the processor in an IP speaker is communicated from its source across a packet-switched data network using IPv4 and IPv6 addressing with a User Datagram Protocol (UDP) or Real-time Transport Protocol (RTP). The IP Speaker connects to Unicast or Multicast addresses to enable the delivery of streamed data from a source on the network, to arrive at a single speaker or many speakers respectively.

See also

Related Research Articles

<span class="mw-page-title-main">Ethernet</span> Computer networking technology

Ethernet is a family of wired computer networking technologies commonly used in local area networks (LAN), metropolitan area networks (MAN) and wide area networks (WAN). It was commercially introduced in 1980 and first standardized in 1983 as IEEE 802.3. Ethernet has since been refined to support higher bit rates, a greater number of nodes, and longer link distances, but retains much backward compatibility. Over time, Ethernet has largely replaced competing wired LAN technologies such as Token Ring, FDDI and ARCNET.

Multiprotocol Label Switching (MPLS) is a routing technique in telecommunications networks that directs data from one node to the next based on labels rather than network addresses. Whereas network addresses identify endpoints the labels identify established paths between endpoints. MPLS can encapsulate packets of various network protocols, hence the multiprotocol component of the name. MPLS supports a range of access technologies, including T1/E1, ATM, Frame Relay, and DSL.

In computer networking, the maximum transmission unit (MTU) is the size of the largest protocol data unit (PDU) that can be communicated in a single network layer transaction. The MTU relates to, but is not identical to the maximum frame size that can be transported on the data link layer, e.g. Ethernet frame.

<span class="mw-page-title-main">OSI model</span> Model of communication of seven abstraction layers

The Open Systems Interconnection model is a conceptual model from the International Organization for Standardization (ISO) that "provides a common basis for the coordination of standards development for the purpose of systems interconnection." In the OSI reference model, the communications between a computing system are split into seven different abstraction layers: Physical, Data Link, Network, Transport, Session, Presentation, and Application.

Quality of service (QoS) is the description or measurement of the overall performance of a service, such as a telephony or computer network, or a cloud computing service, particularly the performance seen by the users of the network. To quantitatively measure quality of service, several related aspects of the network service are often considered, such as packet loss, bit rate, throughput, transmission delay, availability, jitter, etc.

The Address Resolution Protocol (ARP) is a communication protocol used for discovering the link layer address, such as a MAC address, associated with a given internet layer address, typically an IPv4 address. This mapping is a critical function in the Internet protocol suite. ARP was defined in 1982 by RFC 826, which is Internet Standard STD 37.

Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls, the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.

A broadcast address is a network address used to transmit to all devices connected to a multiple-access communications network. A message sent to a broadcast address may be received by all network-attached hosts.

<span class="mw-page-title-main">VoIP phone</span> Phone using one or more VoIP technologies

A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).

The Precision Time Protocol (PTP) is a protocol used to synchronize clocks throughout a computer network. On a local area network, it achieves clock accuracy in the sub-microsecond range, making it suitable for measurement and control systems. PTP is employed to synchronize financial transactions, mobile phone tower transmissions, sub-sea acoustic arrays, and networks that require precise timing but lack access to satellite navigation signals.

IP multicast is a method of sending Internet Protocol (IP) datagrams to a group of interested receivers in a single transmission. It is the IP-specific form of multicast and is used for streaming media and other network applications. It uses specially reserved multicast address blocks in IPv4 and IPv6.

In computer networking, jumbo frames are Ethernet frames with more than 1500 bytes of payload, the limit set by the IEEE 802.3 standard. The payload limit for jumbo frames is variable: while 9000 bytes is the most commonly used limit, smaller and larger limits exist. Many Gigabit Ethernet switches and Gigabit Ethernet network interface controllers and some Fast Ethernet switches and Fast Ethernet network interface cards can support jumbo frames.

<span class="mw-page-title-main">Computer network</span> Network that allows computers to share resources and communicate with each other

A computer network is a set of computers sharing resources located on or provided by network nodes. Computers use common communication protocols over digital interconnections to communicate with each other. These interconnections are made up of telecommunication network technologies based on physically wired, optical, and wireless radio-frequency methods that may be arranged in a variety of network topologies.

In audio and broadcast engineering, Audio over Ethernet is the use of an Ethernet-based network to distribute real-time digital audio. AoE replaces bulky snake cables or audio-specific installed low-voltage wiring with standard network structured cabling in a facility. AoE provides a reliable backbone for any audio application, such as for large-scale sound reinforcement in stadiums, airports and convention centers, multiple studios or stages.

Mobile VoIP or simply mVoIP is an extension of mobility to a voice over IP network. Two types of communication are generally supported: cordless telephones using DECT or PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G or 4G protocols.

Voice over Internet Protocol (VoIP) recording is a subset of telephone recording or voice logging, first used by call centers and now being used by all types of businesses. There are many reasons for recording voice over IP call traffic such as: reducing company vulnerability to lawsuits by maintaining recorded evidence, complying with telephone call recording laws, increasing security, employee training and performance reviews, enhancing employee control and alignment, verifying data, sharing data as well as customer satisfaction and enhancing call center agent morale.

In computer networking, an Ethernet frame is a data link layer protocol data unit and uses the underlying Ethernet physical layer transport mechanisms. In other words, a data unit on an Ethernet link transports an Ethernet frame as its payload.

Walter David "Dave" Sincoskie was an American computer engineer. Sincoskie installed the first Ethernet local area network at Bellcore, and helped invent voice over IP technology. Sincoskie authored the first local ATM specification. He is also the inventor of the VLAN.

In computer networking, the link layer is the lowest layer in the Internet protocol suite, the networking architecture of the Internet. The link layer is the group of methods and communications protocols confined to the link that a host is physically connected to. The link is the physical and logical network component used to interconnect hosts or nodes in the network and a link protocol is a suite of methods and standards that operate only between adjacent network nodes of a network segment.

AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA, Livewire, Q-LAN and Dante.

References

  1. "Speakers". VoIP Supply. Retrieved 2015-04-23.
  2. John Detreville; W. David Sincoskie (December 1983). "A Distributed Experimental Communications System". IEEE Journal on Selected Areas in Communications. SAC-1 (6): 1070–1075. doi:10.1109/JSAC.1983.1146021. S2CID   21400971.
  3. Polle T. Zellweger; Douglas B. Terry; Daniel C. Swinehart (1988). "An overview of the Etherphone system and its applications". [1988] Proceedings. 2nd IEEE Conference on Computer Workstations. pp. 160–168. doi:10.1109/COMWOR.1988.4813. ISBN   0-8186-0810-2. S2CID   57214369.{{cite book}}: |journal= ignored (help)