Fundamentals of VoIP Monitoring & Troubleshooting Part 2
In Fundamentals of VoIP Monitoring & Troubleshooting Part 1 we spoke about the difficulties with reactively troubleshooting VoIP related problems and how Call Detail Records (CDRs) can be used to fill the gap in time. When an end user experiences a problem it can be mins, hours and even days before you’re notified and the troubleshooting process begins. Visibility into what took place during that call is paramount and the metrics gathered from CDR’s can help.
Let’s start with the following call quality metrics:
Network Jitter: Real-time voice communications over the network are sensitive to delay in packet arrival time or packets arriving out of sequence. Excess jitter results in calls breaking up. Jitter can be reduced to a certain extent by using jitter buffers. Jitter buffers are small buffers that cache packets and provide them to the receiver in sequence and evenly spaced for proper playback. Buffer lengths can be modified; however, if jitter buffer is increased too much then the call will experience an unacceptable delay. Consequently, a reduction in buffer turns results in less delay but more packet loss. Jitter is measured in milliseconds (ms).
Latency: Latency, or lag, is the time delay caused in the transmission of a voice packet. Excess latency results in delay, packet drops, and finally to echo. Latency is measured in milliseconds (ms)
Packet Loss: Packet loss occurs when one or more packets of data fail to reach their destination. A single packet loss is referred to as “packet gap”, and series of packet loss is known as” burst”. Packet loss can occur for a variety of reasons including link failure, high congestion levels, misrouted packets, buffer overflows and a number of other factors. Packet loss causes interrupted playback and degradation in voice quality. Packet loss can be controlled using packet loss concealment techniques within the playback codec.
MOS: Mean Opinion Score is a numerical value to indicate the perceived quality of the call from the user’s perspective of the received call after compression, transmission, and decompression. MOS is a calculation based on the performance of the IP network and is defined in the ITU-T PESQ P.862 standard and is expressed as a single number in the range of 1 to 5, where 1 is lowest perceived quality and 5 is the highest. The above metrics are important to monitor and control in order to keep call quality at an acceptable level. It’s also important to note that the above metrics can vary depending where they’re captured. As a best practice it’s a good idea monitor these metrics end to end within your VoIP network. In our next post we’ll talk about how you can capture these statistics from the perspective that matters most – between two VoIP phones after a failed call.
For more information on monitoring and troubleshooting VoIP please read our white paper here.
