Managing voice quality issues over VoIP phone systems

The quality of Voice over IP (VoIP) has improved considerably from past years, and VoIP is now the dominant technology for corporate phone systems. At the moment corporate VoIP is primarily restricted to the corporation intranet and outside calls are converted to analogue and routed via the standard analogue phone backbone. However the technology and bandwidth is now evolving to the point where peer to peer calls between VoIP phone systems across the internet is becoming viable – in terms of quality, reliability and cost.

VoIP phone systems have very high levels of reliability and the latest generation systems are 99.999%+ reliable – of course provided they are properly configured and managed. And VoIP has already proven its case as a cost saving voice system.

The key challenge to achieving this viability is in managing voice quality. Call delay, variable delay and packet loss are the main factors that impact voice quality in a VoIP system. Taking each of these:

Call delay or constant delay refers to the constant delay that can occur in calls, ie a time lag that remains the same throughout the call. This does not directly affect voice quality, however it does impact the way that people communicate. In the extreme this can lead to an awkward lag in the conversation, or over-talking and can have a considerable effect on the quality and flow of conversation.

Variable delay, which is also referred to as jitter, is when transmitted VoIP packets arrive at the distant end of the call at differing time intervals – ie some of the packets are delayed. This is a normal every-day condition for IP based networks – of course for data packets it has little impact as they are simply reordered and joined to recreate the file.

However it is critically important for voice packets, which have to be reordered and joined to create a continuous and close to real-time stream. Jitter results in choppiness and distortion of the analogue recreation that the listener receives.

There are many causes of jitter, including router congestion, operating over parallel routers, changes in mid-stream in the physical infrastructure pathways between terminal clients, transmission issues, codec issues and processor issues.

Many VoIP systems seek to correct for jitter by buffering the incoming packets. The system holds a number of received packets in short-term memory so that any delayed packets can be inserted back into the stream before it is converted back to the analog voice pattern. If jitter is low then the buffer period can be very short. If jitter in the IP network is high then either the buffer period will need to be increased, or there may be perceptible gaps in the conversation. However, increasing the buffer significantly adds to the constant delay discussed above.

Packet loss is when a transmitted packet is not received at the receiving end. This packet loss can be caused by many factors, particularly line quality. The codecs in VoIP System use complex algorithms to compensate for minor packet loss, however they can not fully regenerate or simulate the actual information contained in the lost packets. Hence this packet loss can result in audible gaps in the analog voice when converted at the distant end of the VoIP phone systems.

And overlying all these issues is the challenge of changing and often transient network conditions, and which may be anywhere down the transmission chain.

In order to manage these issues, it is necessary to understand which one or more of these issues is causing the problem. It’s about knowing your enemy, and protecting the voice from the other applications running on your network. The more sophisticated VoIP phone Systems includes considerable functionality to manage these issues. In addition there are various third party diagnostic software programs which are specifically designed to identify IP network problem areas. The functionality may encompass various techniques, such as creating 3 dimensional network time maps, checking the router configuration, analysing the main network pathway(s) to determine if there are time related congestion issues, checking the software and embedded codecs used in terminal equipment are compliant with current standards, and ensuring that the terminal has the quality and processor capability to match with the overall system.

For Further Information:
http://www.minttec.com/voip/