Being a part of the eminent call center we stay focused on the networks and their ability to handle the real-time data. We have to assess from time to time whether our network is able to handle the real-time protocol (RTP) traffic, to gather highly sensitive data.
Once you decide that the network can handle more, and then you have to evaluate the quality of calls and troubleshoot the voice quality issues.
While troubleshooting real-time data like VoIP, you need to have the right analysis tool and search the granular details of the traffic. You need to have the tools for focusing quickly on telephony calls and quality.
MOS and R-Factor – Automated Measurement of Call Quality
In order to evaluate the relative quality of a call, contact centers use Mean Opinion Score (MOS). For years, Fusion has used it in their landlines, where we ask our agents to listen to telephone calls and we rate the call quality from 1 to 5. If the VoIP calls are above 3.5 rating then we receive few complaints.
Human perception and the mathematical analysis by R – Factor helps to understand the call quality in a better way. Moreover, as the VoIP Codec have become better, it is possible for the calls to have an R-Factor greater than 100.
Jitter is another factor that measures the time delay between each RTP packet. If improperly measured, then it can result in gaps and loss of call data. There are equipments, which comes with jitter buffer to smooth out the gaps and reduces call latency.
Analyzing RTP and RTCP to Search the Issue
When we deal with the issue of troubleshooting a network, one has to get inside the granular details. In case of VoIP, two factors Real-Time Transport Protocol (RTP) and Real-Time Transport Control Protocol Packets (RTCP) are important for solving the problems.
RTP is something more than packaging and delivery of voice data. It is also able to troubleshoot issues like:
- A series of numbers helps to detect lost or out of series packets
- The payload type, assigned by the signaling protocol
- A sync source identifier that identifies the unique RTP stream
- A timestamp sets to figure out the packet arrival rate to the listener and to calculate jitter
RTCP packets are useful in because they report on the various details of call progress. RTCP packets generally captured between two VoIP users, to see how many packets drops.
What Should The VoIP Troubleshooting Tool Contain?
At Fusion, when we plan to troubleshoot the poor VoIP call quality, we employ a tool that can measure overall metrics like MOS and R-Factor, along with latency, jitter, and packet loss. This device helps to capture the RTP streams, and to get call quality information close to that user.
The use of robust VoIP tools at Fusion, go beyond simply reporting the jitter. They are also able to check the independent sources of jitter directly from the RTP stream. We also receive the notifications by an end-device via RTCP, in real-time via RTP stream analysis. This two-way analysis helps to pinpoint where in the network jitter starts to become a problem.
Fusion thinks about their customer and business client so they stick to the right metrics and right analysis tool to make their calling service effective.