VoIP – SIP (Voice over IP – Session Initiation Protocol) simulates the VoIP calls between two OpsRamp gateways in your network. This will help to analyze and identify the call response, quality, jitter, and calculate the Mean Opinion Score (MOS).
MOS grade
MOS grade | QoE evaluation |
---|---|
5 | Excellent |
4 | Good |
3 | Common |
2 | Bad |
1 | Very bad |
Prerequisites
- Enable SIP Agent before configuring the SIP synthetics monitor.
- Keep all SIP-enabled gateways in one client.
Configure Parameters
Enter the follwoing details to configure the RTT synthetic monitor:
Section | Field Name | Description | Values/Example | Field Type |
---|---|---|---|---|
Basic Details | Name | Enter a name for the synthetic monitor. | Text | |
Type | Select the type of synthetic monitor. | SIP | Dropdown | |
SIP Configuration | Call Plan | Enter the IP address/host name of the synthetics. It should be accessible from Gateway.
| Dropdown | |
Locations | Select source location. | Dropdown | ||
Call Duration | Duration that verifies the call quality. Call duration cannot exceed the time configured for Call Frequency. | Dropdown | ||
Codec Format | Codec format that determines the format of the voice packets received during the call. | Dropdown | ||
Frequency | The time interval to monitor the chosen Gateways. | Dropdown |
After configuring, view metrics, and graphical representations for more information.
Graphical metrics representation
![Graphical Representation](https://docsmedia.opsramp.com/screenshots/Monitoring/ConfigureSIP-graphicalrepresentation-SIP-5.5.0.png)
Supported metrics
Metric Name | Units |
---|---|
availability.down.location.count Location Count. Number of locations where the configured host is down | - |
sip.call.latency Latency. Time between the moment a voice packet is transmitted and the moment it reaches its destination | milliseconds |
sip.call.pl Packet loss. Percentage of packets dropped by the network on the way to the destination address | percent |
sip.call.mos Mean Opinion Score (MOS): Measure of voice quality | - |
sip.call.jitter Jitter. Time taken to measure the voice instability | milliseconds |
sip.call.rtt Round trip time. Duration for data to travel to the target destination for the speed test and back | milliseconds |