Using Q-Suite’s Call Monitoring Features to monitor Call Center Representatives to improve Customer Satisfaction and First Call Resolution

Call Monitoring features are a must for any call center.   Call Monitoring has many uses with in a business ranging from employee training to  quality assurance and complaint investigation.

How is Call Monitoring useful for training new customer service representatives? Continue reading Using Q-Suite’s Call Monitoring Features to monitor Call Center Representatives to improve Customer Satisfaction and First Call Resolution

VoIP Essentials: Codecs


When it comes to the contact center industry, there is  a plethora of terms, jargon and concepts that can be new, foreign and confusing. Voice over IP, or more commonly, VoIP, adds even more to our shared vernacular. 


One very important concept that permeates our new digital landscape is codec.   It is important to understand what a codec is and why different codecs are used in different applications.

A codec can simply be thought of as a tool that converts audio and or video from one digital format to another digital format.  When a signal is captured from a camera or mic is it transformed with a codec into a digital form, then stored and transmitted to our screens and audio devices.  For example, a very popular codec is the MPEG family of codecs: MPEG-2 is used to transmit audio and video in digital cable boxes and satellite receivers and  MPEG-3 is the codec that is best know as MP3.

With VoIP there are a few popular codecs that are broadly  used and while they all serve the same purpose, they can be vastly different.  VoIP has special constraints that digital broadcasting and digital audio do not have to constrained by.  VoIP has to transmit the audio back and forth to both ends of the call with in milliseconds and it has to do that using a minimal amount of bandwidth. This means that we need to keep in mind:

  • how much time and CPU resources it takes to run the audio through the codec ?
  • how well does the codec compress the audio?
  • how is  the codec is affected by network latency?


When picking a codec there is trade offs with each one,  as some codecs such as G711 use very little compression and sound excellent but they require more bandwidth to transmit so it is more susceptible to audio problems when the network is saturated or if you’re on a WIFI network.  G729 uses a compression process that is highly effective but that comes at a cost.  Most people would not be able to notice the difference a call encoded with G729 vs G711 but G729’s compression process is patented and any phone using it must have the proper license to use it.  Because of this, it’s impossible to find legal softphones that can do the G729 for free.  Being able to use G729 can make the all the difference when the Internet connect is subpar because of the advanced compression.  I’ve been able to have conversations using G729 over weak WIFI signals when the same soft phone with G711 has been unusable.

Another VoIP codec that is popular is the GSM codec.  It was made popular by the cell phone industry as the codec is not patented and it offered an acceptable balance between network utilization and voice quality.  When possible I recommend staying away from GSM because if you do use it,  it will sound just like your on a cell phone. In general, if you are making VoIP calls for business reasons, you’ll want to avoid this codec for this reason.

When I’m setting up a  contact center or PBX using the Q-Suite platform I always make sure that the prefered codec is G711 on the server side, and I let our clients know that, unless there is high bandwidth costs or limited bandwidth between different locations. I make sure that I disable the other codecs like GSM,G726 and G729 if there are no licenses for that codec.  When there is remote users that may be using questionable networks or a need to conserve network bandwidth, G729 is the codec of choice.