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 currently the dominant technology for corporate phone systems. At the minute corporate VoIP is primarily on a the corporation intranet and outside calls are transformed into analogue and routed via the standard analogue phone backbone. However the technology and bandwidth is currently evolving so much that peer to look calls between VoIP phone systems throughout the internet is becoming viable – in terms of quality, reliability and value.
VoIP phone systems have quite high levels of reliability and also the latest generation systems are 99.999%+ reliable – obviously 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 in 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 a period lag that continues to be the same through the call. This won't directly affect voice quality, nevertheless it does impact exactly how people communicate. In the extreme this leads to an awkward lag inside the conversation, or over-talking and may have a considerable influence on the quality and flow of conversation.
Variable delay, and this is referred to as jitter, happens when transmitted VoIP packets reach the distant end in 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 – obviously for data packets it has little impact since they are simply reordered and joined to recreate the file.
However it can be critically important for voice packets, that have to be reordered and joined to produce a continuous and close to real-time stream. Jitter leads to choppiness and distortion from the analogue recreation the listener receives.
There are numerous causes of jitter, including router congestion, operating over parallel routers, modifications in mid-stream within 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 in order that any delayed packets can be inserted back into the stream before it really is converted time for the analog voice pattern. If jitter is low then the buffer period can be very short. If jitter within the IP network is high then either the buffer period will need to become increased, or there could be perceptible gaps in the conversation. However, increasing the buffer significantly adds to the constant delay discussed above.
Packet loss happens when a transmitted packet 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 pay for minor packet loss, they can not fully regenerate or simulate the particular information contained within the lost packets. Hence this packet loss may lead to audible gaps inside the analog voice when converted on the distant end with the VoIP phone systems.
And overlying every one of these issues could be the challenge of changing and sometimes transient network conditions, and which can be anywhere along the transmission chain.
In order to manage these issues, it's necessary to know which one or more of these issues is causing the issue. It’s about knowing your enemy, and protecting the voice from 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 many third party diagnostic software packages which are specifically designed 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 main network pathway(s) to find out if you will find time related congestion issues, checking the application and embedded codecs used in terminal equipment are compliant with current standards, and ensuring the terminal has the quality and processor capacity to match with the general system.
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