Category Video Conferencing

Looking to Streamline Your Video Communications?

LG has arrived on the videoconferencing scene, with support from LifeSize, a division of Logitech. LG’s executive conferencing system, powered by LifeSize, equips customers with a sleek video communications system and a versatile range of high-quality options. This includes a streamlined approach to HD video and voice communications and increased interoperability, all housed within a 24-inch LCD display.

Product highlights

The executive conferencing system has been designed to allow easy deployment, takes a few minutes to set up and only requires the connection of two cables.

It’s fully integrated with HD video and wideband audio, and features:

  • Point-to-Point calls at 720p30 HD resolution
  • An integrated 720p camera, microphone array and speakers
  • Built-in loudspeakers with Echo Cancellation
  • Dynamic Bandwidth Allocation for automatic control of the screen size based on the users’ network

Users will also benefit from an intuitive LifeSize graphical interface, allowing them to easily switch between HD videoconferencing and PC monitor functionality. The on-screen dashboard navigation includes options for shifting between calls, directories, settings and the main menu.

Interoperability

Broad interoperability with video communications systems and IP/PBX installations drives connections with vendors such as Alcatel-Lucent, Asterisk, Avaya, Broadsoft, Cisco, Microsoft and Shoretel. This readily enables executives, remote office employees and teleworkers to connect face to face with anyone, anywhere.

The executive conferencing system will be available to order later this summer, with shipments ready in the fall.

Contact a AVI-SPL Representative at (866) 559-8197 or click here to request more information. »

VGo Teleconferencing Robot Steals the Show at InfoComm

Our AVI-SPL booth was a hot destination last week at InfoComm. Tons of people stopped by to learn more about our products and services, but some got a little sidetracked on the way to our booth.

The VGo teleconferencing robot steals the show at InfoComm 2010

We blame VGo.

VGo is a four-foot tall teleconferencing breakthrough on wheels. We featured the robot at our booth and it simply stole the show.

The VGo offers two-way video and audio. The user can also control the robot’s movements remotely. Imagine not only being able to see and communicate with others in a teleconference, but also freely moving about the room. With VGo, it’s like you’re actually there.

We expect VGo will revolutionize teleconferencing for small and large business, as well as health care. Just picture a doctor checking on patients remotely, or a manager walking the factory line from the airport.

With VGo, you can truly be two places at once.

Call (866) 559-8197 for more information.

Deliver and Access HD Video – Live or On Demand New from LifeSize

LifeSize Video CenterLifeSize recently announced LifeSize® Video Center, an innovative new video capture and broadcast system that enables high definition (HD) video to be accessible everywhere – live or on demand.

  • Executive and corporate communications: communicate with employees across the organization and around the world
  • Training: provide employees the flexibility to view HD training sessions wherever they are, whenever they are available
  • Education and distance learning: broadcast and record classes at the press of a button. Bring faculty to remote students, without compromises in quality

The LifeSize Video Center appliance is specifically designed to process HD video directly where it is created, harnessing the power of LifeSize 220 series HD video communications systems. The architecture is the industry’s first solution to enable an unprecedented 20 concurrent recordings in HD, 1000 simultaneous live streams and up to 350 simultaneous on-demand streams, all in crisp, 720p30 high definition (HD) video.

Other key features of the LifeSize Video Center include:

  • Push Button Recording: record and auto-publish from LifeSize 220 series HD video communications systems with one button
  • Auto-Publish Capability: organize content automatically via recording keys
  • User Controlled Layout: viewers easily switch between layouts of video and
    presentations, during recording and playback
  • Flexibility: record and stream, either in-call or out-of-call for playback live or on demand
  • Intelligent Management: Web portal content automatically customized based on user permission
  • Administrator Control: set up, access and monitor effectively through an intuitive, centralized user interface (UI)

Click here to view more video conferencing products »

An Introduction to Voice Over IP ( VoIP ) with Biamp

BIAMP
Biamp Systems has had a VoIP solution for use in audio conferencing systems for over 2 years now – this technology isn’t new to us. In the past two years, AVI-SPL has installed our solution in systems that are integrated with Cisco Call Manager 5.0 & higher, as well as SIP systems by Avaya, Shoretel and others.

What is VoIP?

Internet Protocol (IP) is the way you can digitally access and send information from a network. You communicate using IP every time you access the internet. Now you can leverage that in new and cost-effective ways. From audio to control to video, IP enabled products are the future backbone for A/V systems. Lower costs, greater flexibility and better management are all advantages of leveraging existing IT infrastructures. With the introduction of the VoIP-2 card, a SIP compliant two-channel Voice over Internet Protocol interface, Biamp Systems follows our tradition of innovating to better serve our customers.

The VoIP-2 Card allows Biamp’s AudiaFLEX to connect directly to IP-based phone systems. Used in conjunction with AEC-2HD Acoustic Echo Cancellation Cards and TI-2 Telephone Interface Cards, the VoIP-2 Card makes AudiaFLEX the most powerful, flexible and affordable telephone conferencing product available. Up to six VoIP-2 Cards can be installed into a single AudiaFLEX unit.

VoIP Basics

Before diving into technical details, this simple overview diagram is a good start to summarize steps involved in a VoIP call.

1. The voice signal is first encoded into a known compressed audio format, packetized in a real-time protocol and then transmitted over the network.

2. A VoIP protocol takes care of managing the communication session.

3. On the receiving side, data is extracted from packets and the signal is decoded back to analog audio. Success of this process is obviously sensitive to delay and packet loss.

Into into VoIP

What Are Those Voice Over IP Acronyms?

Voice over Internet Protocol (VoIP): Protocol specifically designed for voice transmission over networks (LAN/WAN).

Protocol: Similar to how a language enables communication between people, a protocol defines a set of rules used to control connection, communication and data traffic between different network devices.

Codec: It refers to the software algorithm used to encode the voice signal into a compressed data format optimized for transmission over IP. On the receiving side, signal is decoded back to analog audio. Codec quality obviously affects audio performance.

Session Initiation Protocol (SIP): SIP is a widely used peer-to-peer protocol that allows the set up, modification and tear down of a VoIP communication session. Peers of a SIP session are the User Agent Client (Initiating the call) and User Agent Server (Answering the call). Note that SIP does not handle voice transmission, it only manages the communication.

SIP servers: They include the Proxy, Redirect and Registrar Servers. Their purpose is to provide name resolution, user location and pass on messages to other servers in the network.

SIP addresses: Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to an e-mail address and may be of two types: a user name (sip:support@biamp.com) or an E.164 address (5036417287). VoIP-2 card only supports E.164 address.

Real-time Transport Protocol (RTP): An IP packet format that is used for delivering real time audio/video over the LAN/WAN. Once the VoIP call session initialized by SIP, RTP is the protocol used to transmit the voice data.

Quality of Service (QoS): Mechanism used to prioritize applications, users, data flow by guaranteeing a certain level of performance. QoS is very important in the case of RTP applications such as VoIP where it is used to insure quality of the audio signal.

Domain Name System (DNS): DNS procedures provide translation from human friendly hostnames into IP addresses. The SIP session mainly uses DNS to allow a client to resolve a SIP URI into the IP address, port and transport protocol.

SIP call flow process: During the registration process, SIP devices register to a registrar server their SIP addresses. The network is then aware of the location of a device upon request. When a user initiates a call, the SIP discovery process starts by sending a request to a SIP server (proxy or redirect server). The challenge for the proxy server is to obtain the IP address of the device such that voice data can be routed between them. Negotiating a compatible data format (sample rate, codec …etc) is the next step before voice data can be transmitted between parties. SIP terminates the call session with a BYE message at the end of the call.

Click for featured BIAMP products »

AVI-SPL’s NYC Hall Situation Room Featured on CBS Evening News

In coverage of the recent bomb threat to New York City, anchor Katie Couric interviews Mayor Bloomberg in the high-tech Situation Room. AVI-SPL’s critical control room solution highlights advanced videoconferencing technology, an innovative 1 x 2 video wall, 46″ LCD displays and highly-secure communications for rapid decision making.

From more details on the Situation Room’s technology, including the Crestron programming and IP-based systems, click here.