Constant delay, or, in other words, delay that remains the same throughout the call, does not affect voice quality.  It does affect how the people on the call perceive the end of speaking from the distant end, and the timing of the start of conversation following receiving voice from the distant end.

Variable delay (jitter) is when transmitted VoIP packets arrive at the distant end at differing time intervals.  This condition is a normal, standard, every-day reality in IP networks. 

There are many causes of jitter:  router congestion, parallel router operation, changes in physical pathways between the terminal clients, transmission issues, codec issues, and processor issues.

The result of jitter is choppiness and distortion in the analog recreation of the received voice packets. 

Many VoIP systems correct for jitter by buffering incoming packets.  This means that a short-term memory device holds the received packets for a pre-determined time in order to make sure that enough packets are availed to recreate the analog voice pattern without perceptible gaps in the conversation.

The use of a jitter buffer to correct for variable packet delay automatically creates increased constant delay in the system as the operation of the buffer is a “delay function” to smooth the received packet stream.

To correct for high levels of jitter in the IP network, larger jitter buffers are required.  The larger the buffer, the longer the delay added to the system.

Packet loss is when a transmitted packet is not received at the distant end.  There are many causes of packet loss in an IP system.  While the algorithms used in many codecs can compensate for minor packet loss, no system can fully recreate / simulate the information contained in packets transmitted but not received.  Packet loss means large gaps in the re-created analog voice at the distant end of a VoIP system.

The difficulty in solving these problems is that the network conditions that are the underlying cause of poor voice quality are often transient, and are sometimes in router or transmission locations far away from the VoIP equipment and users.

We have found that Ping Plotter Pro, a diagnostic software program specifically designed to identify IP problem areas, is a great tool in this multi-dimensional search for voice quality.

Other diagnostic techniques include analzing the specific pathway that the VoIP real time stream uses to determine if there is a time-of-day congestion issue, checking the codecs used in terminal equipment for standards compliance (verify the software used, testing the codec itself is far beyond the skill set of the everyday VoIP analyst!), and making sure that the actual digital to analog interface is commensurate with the quality expected in the system (a $10 headset may not recreate a $1,000 waveform successfully).