Posts tagged VOIP

Webinar Recording: VoiP Conferencing, by Biamp

AVI-SPL has made available the recording of “The Best Quality Solutions in Mid- to Large-Room VoIP Conferencing,” a webinar presented by Biamp Systems’ Frank West. Frank, Biamp’s south central regional manager, discussed improvements in audio conferencing, the advantages of a networked audio system, audio video bridging, and Biamp’s Audia and Tesira solutions.

On the Agenda

  • How VoIP improves productivity
  • Developing a better VoIP conference experience
  • Why companies rely on VoIP room solutions

Play the webinar “The Best Quality Solutions in Mid- to Large-Room VoIP Conferencing.

Register for August 21 Webinar on VoIP Conferencing by Biamp

Join us Tuesday, August 21 at 1 p.m. EST for “The Best Quality Solutions in Mid- to Large-Room VoIP Conferencing,” presented by Biamp Systems’ Frank West. Frank will discuss five key reasons why companies have deployed VoIP room conferencing systems to improve their productivity. He will also discuss the steps needed to develop a better VoIP conferencing experience.

On the Agenda

  • How VoIP improve productivity
  • Developing a better VoIP conference experience
  • Why companies rely on VoIP room solutions

Register for “The Best Quality Solutions in Mid- to Large-Room VoIP Conferencing.”

About the Presenter

Frank West is the South Central Regional Manager for Biamp Systems, an internationally recognized leader in the professional audio conferencing industry. Frank has over 15 years of experience in the professional AV industry. Over the last five years, Frank has implemented thousands of VoIP end points with multiple microphone inputs to provide a better end user VoIP experience.

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.

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Study: Unified Communications Provides 4X the ROI

Looking for proof that advanced Unified Communications and Collaboration (UC&C) technology is a worthy investment for your company? Just turn to the latest report from business research and consulting firm Frost & Sullivan. Sponsored by Verizon and Cisco, “Meetings Around the World II: Charting the Course of Advanced Collaboration,” introduces the industry’s first quantitative model for a return on collaboration investment.

Excerpts from the study:

  • Businesses and government agencies deploying VoIP soft phones, immersive video and fixed mobile convergence received an average return of four times their investment in deploying collaboration technologies in terms of improvement across business-critical areas.
  • The majority of organizations deploying (UC&C) are more successful than their peers compared to those not deploying UC&C (70 percent versus 47 percent).
  • Of organizations deploying UC&C, 40 percent plan to increase spending.
  • More than 80 percent of organizations that have not yet deployed UC&C tools plan to deploy some form in the next two to three years.

Based on this research, are you interested in developing a UC & C integration for your organization? If not, what are the key factors holding you back?

For more on the study, click here.

AMX Transforms Intercom-Enabled Touch Panels Into VoIP Phones

Users Can Make or Receive Telephone Calls from Select Modero Touch Panels

AMX®, a leader in providing simplified solutions in the audio/visual control industry, unveils its Telephone/Intercom Integration Package, enabling users to make and receive local, long distance and international calls from selected Modero® touch panels with the AMX SIP Gateway.

AMX has added new telephone features including Caller ID, Call waiting, Call groups, one-touch dialing, and voicemail, as well as access to local and shared address books from the touch panel and SIP Gateway, respectively. Customers are welcome to use their existing telephone service and simply add the SIP Gateway to take advantage of these enhanced calling features. The technology is seamless to the customer and is as simple as pressing an on-screen icon to initiate or receive a call. The enhanced touch panels also deliver PBX functionality to homes or small offices and allow telephone access via VoIP and/or analog telephone lines.

Select Modero touch panels now support full motion JPEG offering at least 24 frames per second and enhanced scalability of full motion JPEG video. The SIP Gateway supports up to 50 telephony devices including analog phones, SIP phones and these selected Modero intercom-enabled touch panels:

  • 5.2″ Modero ViewPoint Touch Panel with Intercom (MVP-5200i)
  • 8.4″ Modero ViewPoint Touch Panel with Intercom (MVP-8400i)
  • 10″ Modero Wall/Flush Mount Touch Panel with Intercom (NXD-1000Vi)
  • 7″ Modero Wall/Flush Mount Touch Panel with Intercom (NXD-700Vi)
  • 5″ Modero Wall/Flush Mount Touch Panel (NXD-500i)

Click here to view our line-up of AMX products. »