Using Voice Test Views

Give your hybrid team and customers a superior collaboration experience with clear, reliable VoIP calls. Watch this tutorial to learn how to use the RTP stream and SIP server tests within the ThousandEyes platform to access a comprehensive view of your organization’s voice quality, helping you identify and resolve issues quickly.

We’ll cover the RTP stream test’s View page, the SIP server test’s View page, and how to navigate them to get the needed insights.

More About the RTP Stream Test
The RTP stream test creates a direct connection between the VoIP caller and receiver, sending voice packets to measure aspects like Mean Opinion Score (MOS), packet loss, discards, latency, and Packet Delay Variation (PDV). It also lets you configure DSCP and codec values.

You can use the test to:

• Identify the node responsible for the degraded RTP stream
• Analyze the effects of BGP route changes on the RTP strea
• Measure call audio quality

More About the SIP Server Test
SIP server monitoring helps you ensure optimal call quality and prevent service disruptions. The ThousandEyes SIP server test provides real-time performance insights to enable you to detect problems early and deliver consistent, high-quality VoIP services.

The test measures the following key metrics:

• Availability: This measure refers to the percentage of time that the SIP server provides a successful response. Availability tests the basic connectivity to and the response from the target SIP server.
• Response Time: Also known as time-to-first-byte, response time refers to the time elapsed from the request initiation until the first byte of server response is received.
• Total Time: This metric represents the time taken to complete all phases of the test. It provides a detailed breakdown of timings per agent and overall averages.

Give your hybrid team and customers a superior collaboration experience with clear, reliable VoIP calls. Watch this tutorial to learn how to use the RTP stream and SIP server tests within the ThousandEyes platform to access a comprehensive view of your organization’s voice quality, helping you identify and resolve issues quickly.

We’ll cover the RTP stream test’s View page, the SIP server test’s View page, and how to navigate them to get the needed insights.

More About the RTP Stream Test
The RTP stream test creates a direct connection between the VoIP caller and receiver, sending voice packets to measure aspects like Mean Opinion Score (MOS), packet loss, discards, latency, and Packet Delay Variation (PDV). It also lets you configure DSCP and codec values.

You can use the test to:

• Identify the node responsible for the degraded RTP stream
• Analyze the effects of BGP route changes on the RTP strea
• Measure call audio quality

More About the SIP Server Test
SIP server monitoring helps you ensure optimal call quality and prevent service disruptions. The ThousandEyes SIP server test provides real-time performance insights to enable you to detect problems early and deliver consistent, high-quality VoIP services.

The test measures the following key metrics:

• Availability: This measure refers to the percentage of time that the SIP server provides a successful response. Availability tests the basic connectivity to and the response from the target SIP server.
• Response Time: Also known as time-to-first-byte, response time refers to the time elapsed from the request initiation until the first byte of server response is received.
• Total Time: This metric represents the time taken to complete all phases of the test. It provides a detailed breakdown of timings per agent and overall averages.