Now Available From Grandstream Networks

Grandstream Networks, since 2002, has been successfully deploying VoIP solutions to customers across the globe. Ranging from Corded IP Phones to PBX Appliances, there has been almost no VoIP stone that has been unturned by Grandstream in its endeavor to provide the best and most cost effective VoIP solution on the market – that is, until today.

This morning Grandstream released the GAC2500 Conference Phone, the first conference unit ever to be produced by the company. Utilizing the power of Android 4.4 , as well as the many other high-end functions associated with today’s modern VoIP solutions,  the GAC2500 easily delivers a full and robust conference solution for any size business.

What It’s All About

This circular-shaped conference phone features both a modern design, and also many high performance functionalities such as:

• Auto-sensing Gigabit Ethernet with PoE
• Integrated Bluetooth 4.0
• Android 4.4 – allowing for full access to Google Play Store and the potential for custom app creation
• Supports 6 lines with up to 6 SIP accounts
• Features a 4.3″ color touch screen for each navigation and utilization
• Complimentary solution for the GVC3200 Video Conference System, allowing for full utilization of apps such as Skype for Business and Google Hangouts

This product also delivers HD quality audio via 3 cardioid microphone that allow for a pickup range of up to 12 feet. The GAC2500 not only delivers high-quality audio, but also features auto provisioning to ensure an easy deployment, as well as a firmware upgrades via TFTP, HTTP, and HTTPS or via local HTTP upload. The GAC2500 also contains advanced security protection to ensure privacy, as well as broad interoperability with many 3rd party SIP devices.

The GAC2500, at every angle, delivers a very powerful conference solution. Included with the unit is the GAC2500 phone, a universal power supply, a network cable, a USB cable, a RJ48 cascade cable (for daisy-chaining), as well as a Quick Installation Guide.

Where To Find It

VoIP Supply is thrilled to offer this product as the latest and greatest solution from Grandstream Networks,  and would love to hear what you think about this product as well! Please leave a comment or question for us in the below comment area, or contact us directly at 1-800-398-8647 and we’d be happy to tell you more about this solution from Grandstream.

VoIP Supply also thanks Ying-Hui Chen for her assistance with posting this article.

The advantages of Primary Rate Interface (PRI) service and Session Initiation Protocol (SIP) can be debated.

Both protocols use digital signals but the latter provides an advantage that allows you to customize the phone system according to your business needs.

However, it helps to educate yourself on both options before making a decision on which one is best for you.

Primary Rate Interface

What Is PRI?

PRI (Primary Rate Interface) is a interface standard associated with transmission of voice data over a dedicated line.

PRI uses a circuit switched model for voice call among people located in different locations.

How Does It Work?

PRI provides 23 voice channels that a business can use simultaneously at any given time.

Multiple users can access these 23 channels within a specified network and the hardware requirement includes a PBX (Private Branch Exchange) or IP PBXs.

However, if a business has 50 operational DID (Direct Inward Dialing) numbers, then only 23 will be available any point of time. To make all the numbers available more PRI circuits are required (which will increase your overhead).

Support Framework for PRI

Integrated Services Digital Network (ISDN) assigns the PRI service.

ISDN enables the digital transmission of voice and data service over Public Switched Telephone Network (PSTN). The PSTN is where all the global telecommunication devices are connected.

You can call to any part of the world, regardless of the receiver’s device, through the PSTN. Initially, the PSTN provided analog transmission but eventually it was digitized after the advent of ISDN to provide a faster, better quality, and lower cost service.

Session Initiation Protocol

What is SIP?

Session Initiation Protocol or SIP Trunking is a system where transmission of voice data is performed over an existing network data line by creating a virtual network between the PBX and the SIP network of the carrier.

In other words, it goes over an IP Network. It uses a Packet switched model for voice calling.

How Does It Work?

The system uses an IP PBX system rather than the normal PBX (unless it’s compatible with SIP framework). The primary reason for this is the use of VoIP rather than PSTN. The public internet and LAN is connected with an IP PBX.

The voice call in a SIP framework uses the same path allotted for non-voice transmission.

Similarly, when receiving voice data, the equipment at the service provider’s end transmits it to the designated IP PBX. Re-routing of data packets to another pre-determined number takes place when the primary number is not responding.

Support Framework for SIP

SIP uses your existing data transmission set-up. Unlike PRI, SIP does not come in a pre-determined connection bundle.

Depending on your requirements, a business can buy call paths or increase bandwidth of the existing data line.

With SIP, businesses are not dependent upon on a bundle service such as PRI. If the number of calls is less than the actual capacity, the unused bandwidth is distributed across the network for other use. This ensures a more efficient network with no wasted resources.

Note: This is a guest post by Victoria Zona, Senior Channel Sales Manager, Revolabs

Distance learning has become increasingly pervasive in both corporate America and at educational institutions.

Academic institutions are all adding programs to accommodate remote students and are now offering on-line degrees.

To save money, many corporations are remotely training branch office employees on the policies and procedures bringing distance learning to corporate workplaces.

Clear Audio Maintains Focus, Increases Comprehension, Reduces Fatigue

Many times the voice quality in remote learning environments is unintelligible. Poor sound quality makes it difficult for the student to understand the material presented, and reduces their active participation.

Our industry has made great strides to improve video quality during distance learning sessions, but crystal-clear audio is even more important to keep students’ focus, increase comprehension, and prevent classroom fatigue.

Typically, there’s no technical person on hand to monitor and support the remote learning session. Therefore, having an audio solution that is simple to use and easy to setup is key.

Poor sound quality makes it difficult for the student to understand the material presented, and reduces their active participation.

How to Choose Audio Devices for Distance Learning

Here are some tips to consider when choosing an audio product for distant learning:

Seek out an audio solution that provides superior audio pickup and speaker quality so users on both the near and far end of the room can hear every word.

360° Coverage

Deploy a device that has full 360° coverage of audio capture, incorporates integrated acoustic echo cancellation, and supports full duplex audio to enhance comprehension and conversational flow.

Strong Speaker, External Connections

The speaker output on the device needs to be loud enough for the participants in the room to hear, or allows connections to external speakers to amplify the sound level in the room.

BYOD Compatible

If bring your own device (BYOD) program is supported, make sure the audio conferencing device is software-agnostic and connects via standard PC, tablet or mobile communications channels.

Cloud Compatibility

For cloud based conferencing solutions, verify that the audio conferencing device supports multiple communication applications.

Ask a VoIP Supply Representative about Revolabs audio conferencing solutions for distance learning.

We have a range of products to fit every environment, offering unmatched audio quality that allows users to hear every word.
 

Author Bio

Victoria Zona
Senior Channel Sales Manager, North America, Revolabs
Victoria Zona has more than 20 years of experience in channel sales. In addition, she was also recognized as a CRN Channel Chief for 2014. At Revolabs she is responsible for directing and managing the North America DMR channel. Connect with Victoria on LinkedIn and Twitter