Managing voice quality issues over VoIP phone systems
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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 limited to the corporation intranet and outside calls are transformed into analogue and routed through the standard analogue phone backbone. However the technology and bandwidth is evolving to the point where peer to look calls between VoIP phone systems through the internet has become viable – when it comes to quality, reliability and cost.
VoIP phone systems have quite high levels of reliability as well as the latest generation systems are 99.999%+ reliable – needless to say provided they are properly configured and managed. And VoIP has proven its case being a cost saving voice system.
The key challenge to accomplishing this viability is within managing voice quality. Call delay, variable delay and packet loss would be the main factors that impact voice quality in a very VoIP system. Taking all these:
Call delay or constant delay refers to the constant delay that may occur in calls, ie an occasion lag that continues to be the same through the call. This won't directly affect voice quality, nevertheless it does impact the way that people communicate. In the extreme this leads to an awkward lag within the conversation, or over-talking and may have a considerable relation to the quality and flow of conversation.
Variable delay, also is referred to as jitter, happens when transmitted VoIP packets reach the distant end in the call at differing time intervals – ie some from the packets are delayed. This can be a normal every-day condition for IP based networks – obviously for data packets it has little impact since 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 make a continuous and near to real-time stream. Jitter brings about choppiness and distortion from the analogue recreation the listener receives.
There a wide range of causes of jitter, including router congestion, operating over parallel routers, alterations in mid-stream inside the physical infrastructure pathways between terminal clients, transmission issues, codec issues and processor issues.
Many VoIP systems aim to correct for jitter by buffering the incoming packets. The system holds several received packets in short-term memory to ensure that any delayed packets might be inserted back into the stream before it's converted returning to the analog voice pattern. If jitter is low then a buffer period may be very short. If jitter inside the IP network is high then either the buffer period will need to be increased, or there might be perceptible gaps inside conversation. However, enhancing the buffer significantly enhances the constant delay discussed above.
Packet loss happens when a transmitted packet is not received on the receiving end. This packet loss may be caused by many factors, particularly line quality. The codecs in VoIP System use complex algorithms to pay for minor packet loss, nevertheless they can not fully regenerate or simulate the specific information contained in the lost packets. Hence this packet loss may result in audible gaps within the analog voice when converted on the distant end with the VoIP phone systems.
And overlying every one of these issues will be the challenge of changing and quite often transient network conditions, and that could be anywhere down the transmission chain.
In order to control these issues, it can be necessary to be aware of which one or even more of these problems is causing the issue. It’s about knowing your enemy, and protecting the voice in the other applications running on your own network. The more sophisticated VoIP phone Systems includes considerable functionality to deal with these issues. In addition there are several third party diagnostic applications which are created specifically to identify IP network problem areas. The functionality may encompass various techniques, including creating 3 dimensional network time maps, checking the router configuration, analysing the primary network pathway(s) to find out if you can find time related congestion issues, checking the program and embedded codecs employed in terminal equipment are compliant with current standards, and ensuring that the terminal has the product quality and processor capacity to match with the overall system.
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