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 minute corporate VoIP is primarily tied 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 currently evolving enough where peer to look calls between VoIP phone systems over the internet is now viable – in terms of quality, reliability and price.
VoIP phone systems have very high levels of reliability and the latest generation systems are 99.999%+ reliable – naturally provided they are properly configured and managed. And VoIP has already proven its case like a cost saving voice system.
The key challenge to accomplishing this viability is within managing voice quality. Call delay, variable delay and packet loss will be the main factors that impact voice quality in a very VoIP system. Taking each one of these:
Call delay or constant delay refers to the constant delay that may occur in calls, ie a period lag that is still the same through the entire call. This does not directly affect voice quality, nevertheless it does impact the way that people communicate. In the extreme this leads to an awkward lag inside conversation, or over-talking and will have a considerable influence on the quality and flow of conversation.
Variable delay, which is also referred to as jitter, happens when transmitted VoIP packets arrive at the distant end with the call at differing time intervals – ie some with the packets are delayed. This can be a normal every-day condition for IP based networks – needless to say for data packets it's got little impact because they are simply reordered and joined to recreate the file.
However it's critically important for voice packets, that have to be reordered and joined to generate a continuous and near to real-time stream. Jitter leads to choppiness and distortion from the analogue recreation how the listener receives.
There are numerous causes of jitter, including router congestion, operating over parallel routers, modifications in mid-stream in the physical infrastructure pathways between terminal clients, transmission issues, codec issues and processor issues.
Many VoIP systems look to correct for jitter by buffering the incoming packets. The system holds numerous received packets in short-term memory to ensure any delayed packets might be inserted back into the stream before it really is converted back to the analog voice pattern. If jitter is low then this buffer period may be very short. If jitter inside IP network is high then either the buffer period will need to be increased, or there could be perceptible gaps inside the conversation. However, enhancing the buffer significantly increases the constant delay discussed above.
Packet loss is the place a transmitted packet isn't received on the receiving end. This packet loss can be caused by many factors, particularly line quality. The codecs in VoIP System use complex algorithms to pay for minor packet loss, however they can not fully regenerate or simulate your information contained inside lost packets. Hence this packet loss can lead to audible gaps in the analog voice when converted at the distant end in the VoIP phone systems.
And overlying each one of these issues may be the challenge of changing and sometimes transient network conditions, and which may be anywhere along the transmission chain.
In order to control these issues, it can be necessary to understand which one or even more of these problems is causing the challenge. It’s about knowing your enemy, and protecting the voice through the other applications running on your network. The more sophisticated VoIP phone Systems includes considerable functionality to handle these issues. In addition there are numerous third party diagnostic applications which are specifically designed to identify IP network trouble spots. 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 there are time related congestion issues, checking the application and embedded codecs employed in terminal equipment are compliant with current standards, and ensuring that this terminal has the high quality and processor capability to match with the entire system.
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