Managing voice quality issues over VoIP phone systems
Managing voice quality issues over VoIP phone systems
The quality of Voice over IP (VoIP) has improved considerably from past years, and VoIP is the dominant technology for corporate phone systems. At the second corporate VoIP is primarily tied to the corporation intranet and outside calls are converted to analogue and routed through standard analogue phone backbone. However the technology and bandwidth is evolving enough where peer to see calls between VoIP phone systems across the internet is starting to become viable – in terms of quality, reliability and cost.
VoIP phone systems have high levels of reliability along with 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 achieving this viability is at managing voice quality. Call delay, variable delay and packet loss are the main factors that impact voice quality inside a VoIP system. Taking each one of these:
Call delay or constant delay refers back to the constant delay that can occur in calls, ie a moment lag that continues to be the same through the entire call. This will not directly affect voice quality, nonetheless it does impact how people communicate. In the extreme this may lead to an awkward lag within the conversation, or over-talking and may have a considerable effect on the quality and flow of conversation.
Variable delay, also is referred to as jitter, happens when transmitted VoIP packets reach 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 – of course for data packets it's little impact because they are simply reordered and joined to recreate the file.
However it can be critically important for voice packets, which may have to be reordered and joined to make a continuous and all-around real-time stream. Jitter brings about choppiness and distortion with the analogue recreation that this listener receives.
There are many 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 numerous received packets in short-term memory in order that any delayed packets could be inserted back into the stream before it is converted returning to the analog voice pattern. If jitter is low then your buffer period may be very short. If jitter in the IP network is high then either the buffer period will need to be increased, or there could be perceptible gaps in the conversation. However, helping the buffer significantly adds to the constant delay discussed above.
Packet loss occurs when a transmitted packet just isn't received with the receiving end. This packet loss could be caused by many factors, particularly line quality. The codecs in VoIP System use complex algorithms to make up for minor packet loss, nevertheless they can not fully regenerate or simulate your information contained inside lost packets. Hence this packet loss can lead to audible gaps within the analog voice when converted with the distant end with the VoIP phone systems.
And overlying every one of these issues may be the challenge of changing and frequently transient network conditions, and which might be anywhere along the transmission chain.
In order to handle these issues, it really is necessary to be aware of which one or more of these complaints is causing the issue. It’s about knowing your enemy, and protecting the voice from your other applications running on your own network. The more sophisticated VoIP phone Systems includes considerable functionality to control these issues. In addition there are numerous third party diagnostic applications which are specifically made to identify IP network troublesome areas. The functionality may encompass various techniques, including creating 3 dimensional network time maps, checking the router configuration, analysing the key network pathway(s) to ascertain if there are time related congestion issues, checking the software and embedded codecs utilized in terminal equipment are compliant with current standards, and ensuring that the terminal has the product quality and processor capacity to match with the entire system.
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