Some of the wireless service providers use an audio conference bridge in conjunction with such drive testing. A standard test signal is fed into an audio conference that is set up for ‘lecture mode’. In this mode the incoming signal is simply broadcast downstream. Different testers then dial into the conference using their cell phones and hear the test signal as they drive around. Since the test signal is a continuous tone, it helps them characterise where the signal becomes weak or is completely absent. They can also visit the spots where consumers complain about dropped calls and verify if that happens due to a weak signal or due to some other reason.

Modern audio conference bridges allow one to see a real time view of a running conference. Based on different thresholds for loudest speaker algorithm, the test supervisor can easily tell which routes have signal strength issues.

This approach of using a conference bridge for wireless signal strength testing can be cost effective and add a human angle to the test results.

Another side benefit of the real time display of the conference is that the test supervisor can tell which testers are dialed into the bridge and which ones are goofing off.

That’s pathetic, because audio conference calls are the most underused but highest value tool for helping a distributed team work together. While we are continuing to see many advances in desktop videoconferencing and Web-based conferencing, the fact is that audio conference calls are the simplest, cheapest, easiest way to bring a group together.

Modern bridges provide a number of features such as mute/unmute, mute all/ unmute all, self mute/ self un mute, enter conference muted, lock/ unlock, loudest speaker display, transfer to private room etc., that can make your conferencing experience pleasant. Some conference call bridges allow use of these functions via use of DTMF keys or via a web portal or both.

Next time you get on a conference call, ask the conference co-ordinator about what in-conference controls are available to you.

Have your ever thought about the command and control infra structure that must be in place at NASA to pull off such feats?

This is where mission critical conferencing plays an important part.

Typically, different groups of engineers need to communicate with each other on different conferences. For example, a group may be discussing ‘fuel utilization’ and another group ‘propulsion parameters’ etc.

There are several supervisors who also need to keep track of their sub ordinates and speak to their peers at moment’s notice. Some of these conversations may be point to point or on a conference call.

The mission director needs to be able to listen into multiple conversations that may be going on in different conference rooms.

Based on what the mission controller hears, he/she may have to barge in and have full duplex conversation with people in a given conference room.

All conferences taking place need to be recorded for future reference.

The satellite launches are typically monitored from different locations around the world. The people in these locations need to be able to talk to each other in ‘always on’ conference calls.

These multiple conference calls need to occur in a secure environment.

The conference bridge equipment should be able to display real time call activity, including display of loudest speaker etc.

A number of in conference call controls e.g., mute/ un-mute, lock/unlock etc should be available and easily usable to keep the conference calling experience user friendly.

All of the capabilities described above should be accessible from a command and control web portal.

Together this entire arrangement is called Mission Critical Audio Conferencing.

Following definitions are available on Wikipedia and elsewhere on the internet for these terms.

The Joint Interoperability Test Command (JITC) is a US military organization that tests technology that pertains to multiple branches of the armed services and government. JITC’s mission is to provide a full-range of rapid, standardized and customized test, evaluation, and certification services to support global net-centric war fighting capabilities under all conditions of peace and war.

The Defense Switched Network (DSN) is a primary information transfer network for the Defense Information Systems Network (DISN). The DSN provides the worldwide non-secure voice, secure voice, data, facsimile, and video teleconferencing services for DOD Command and Control (C2) elements, their supporting activities engaged in logistics, personnel, engineering, and intelligence, as well as other Federal agencies.

The Unified Capabilities Certification Office (UCCO) is a government organization that serves as a central coordinator for JITC testing and approval. It provides Sponsor/vendor tracking numbers to track product, manages IA and IO test schedules, coordinates and tracks product status on testing schedule, submits the proper certification documentation for the product to the DISN Security Accreditation Working Group (DSAWG), and contacts the sponsor with the decision regarding their submittal.

The DISA Security Technical Implementation Guides (STIG) define the technical security policies, requirements, and Implementation details for applying security to the DISN.

The Unified Capabilities Requirements (UCR 2008) specifies technical standards for telecommunication switching equipment to be connected to the DISN; the emphasis is on Military Unique Features, e.g., Multilevel Precedence and Preemption (MLPP).

The Information Assurance (IA) is the practice of managing risks related to the use, processing, storage, and transmission of information or data and the systems and processes used for those purposes. JITC related IA testing is typically Composed of two (2) phases; Phase I: Security Technical Implementation Guide (STIG) compliance and Functional Security Tests and Phase II: IP Penetration Testing and Telephony Testing.

The Inter Operability (IO) refers to a product being able to connect to the DSN and provide needed telecommunications functions as per UCCO defined STIG documents.

Managed Service: This refers to company buying audio conferencing service from a service provider. Most of the phone companies , large and small offer audio conferencing service these days. These companies then bill based on number of minutes used. The rates for conference calls range from 5 – 20 cents per minute based on number of minutes used. Typically the phone company also provides an 800 number for using the service. So the charge for long distance call is built into the per minute price.

On Premises Conference Bridge: This refers to a company deploying a conference bridge in its own network. The bridge is usually deployed behind a company’s PBX. A TIE trunk is required between the PBX and the bridge. Depending on the PBX and the conference bridge, the TIE trunk can be either based on TDM connection (CAS or PRI T1 or E1) or based on SIP/ H.323 connection. A on premises bridge makes sense when monthly conference usage goes above a few hundred Dollars per month. The conference bridges are sold on a per port basis. One port equates to one voice call. So a 24 port conference bridge can support up to 24 callers. These callers can be in one 24 party conference or four 6 party conferences or any combination in between. 

Owning a conference call bridge also means that you have to maintain it. Modern conference bridges provide web portals for this purpose and are usually easy to learn and use.

XOP Networks has extended the ring down bridging concept to VoIP based networks. It’s  <strong>SIP/VoIP based Ring Down bridge</strong> (Enhanced Firebar application running on SIP based USN) works in conjunction with an IP PBX. A person can go off-hook on pre-designated IP extension. The IP PBX will be programmed to send a SIP INVITE (equivalent to seeing an off-hook in the analog world) towards the XOP bridge. Based on that message the XOP equipment will immediately ring other pre-designated IP phones and upon off-hook place people into an audio conference.