Voice Tests
Voice tests provide a traceroute-like visualization of the network route and can optionally provide the BGP Route Visualization view showing a map of connections between autonomous systems in the route from each BGP monitor to the target.
Related Video
Configuring Voice Tests
Watch this video for a walkthrough of configuring voice tests.
SIP Server Test
A SIP server test checks the availability of Session Initiation Protocol VoIP Server. Additionally, this test can be configured to perform a SIP registration with a target SIP server. Optionally it can verify response status code and whether response headers match a configured regular expression. For more details, see Using the SIP Server View.
Typical Use Cases
Verifying performance of a VoIP SIP server.
Confirming the ability to perform SIP Register with a target server.
Identify the phase at which a SIP Register fails.
Observe the SIP Register and Options request and response.
Example SIP Server Test Results
Here is a SIP Server test targeting an internal VoIP server.
Other Included Tests
Agent-to-server test
Manually Configuring SIP Server Tests
To configure a SIP Server test:
In the ThousandEyes platform, navigate to Network & App Synthetics > Test Settings.
Click the Add New Test button in the upper right corner of the screen.
From the menu that appears, select the card titled SIP Server, located on the bottom left.
The configuration page is divided into the following sections: Basic Settings, Server Configuration and Network Settings, and Additional Settings.
SIP Server Test Basic Settings
SIP Server
The hostname or IP address of the SIP server to test.
Test name (optional)
A name for the test. This auto-populates with the SIP server and port number but can be manually edited.
How often test runs
The frequency at which the test runs. Defaults to 2 minutes. More frequent intervals are recommended for critical services.
Where test runs from
Select one or more ThousandEyes Agents to run the test from. Agents act as observation points to collect performance data.
SIP Proxy
When enabled, allows you to measure the performance and availability of a SIP proxy to detect routing issues or delays. Disabled by default.
Protocol
The transport protocol used for SIP signaling.
TCP: (Default) Use when SIP signaling messages or reliable, ordered delivery are needed, especially across intermediate servers or stricter firewalls.
UDP: Use for lightweight SIP OPTIONS reachability checks or REGISTER requests on stable, low-loss networks when encryption is not required.
TLS: Use when encrypted SIP signaling is required.
Port
The destination port on the SIP server. Defaults to 5060 for TCP or UDP and 5061 for TLS.
Alerts
Enable or disable alerts for this test and assign one or more alert rules. Enabled by default; default alert rules are selected when available.
Tags
Assign tags to this test for filtering and organization. No tags are assigned by default.
SIP Server Test Server Configuration and Network Settings
Test execution schedule
Determines the timing of test rounds for all assigned agents.
Default: All agents start the test simultaneously.
Randomized Start Time: Staggers agent start times randomly within the test interval.
Timeout
Defines the maximum time (default 5 seconds) to wait for a server response. Adjust this based on expected server performance.
Target response time
Sets a performance threshold for the server's response time. Defaults to 1000 ms. Responses exceeding this value will be flagged with a warning.
Desired status code
Verifies the SIP response status code. Use the default 2xx or 3xx response range, or select Other to enter a specific SIP status code from 100 through 606.
Verify headers
When enabled, checks SIP response headers for accuracy to detect misrouting or misconfigurations. Disabled by default.
Perform SIP Register
When enabled, the test performs a SIP REGISTER request to verify credentials and registration status. Disabled by default.
Username
The SIP username to use for registration. This field appears when Perform SIP Register is enabled.
Custom authentication username
When enabled, lets you enter a different authentication username than the SIP username used for registration. This field appears when Perform SIP Register is enabled.
Password
The password to use for SIP registration. This field appears when Perform SIP Register is enabled.
Perform Network Measurements
When enabled, the test also collects underlying network performance data (loss, latency, jitter) to the SIP server. Enabled by default.
Define which data to collect
Select the types of network data to collect. Maximum Transmission Unit (MTU) and Collect BGP data are selected by default.
Maximum Transmission Unit (MTU): Determines the maximum packet size that can be sent without fragmentation.
Collect BGP data: Correlates network data with BGP routing information.
Transmission rate
Controls the rate at which network measurement packets are sent. Defaults to Not fixed. Use to simulate network load or validate QoS policies.
Number of path traces
Sets the number of path traces performed in each test round to help discover multiple network paths. Defaults to 3. Custom values can range from 3 to 10.
Path trace mode
When In session is selected, performs the path trace within an established TCP session to avoid issues with stateful firewalls. Defaults to In session for TCP and TLS SIP tests. This field appears for TCP and TLS SIP tests, but not UDP SIP tests.
Probing mode
Defines the TCP probing method for network measurements. This field appears for TCP and TLS SIP tests, but not UDP SIP tests.
Prefer SACK: (Default) Uses a single TCP connection for efficient measurement.
Force SACK: Forces SACK-based measurements.
Force SYN: Sends a stream of SYN packets; useful if SACK is not supported.
IPv6 policy
Determines how the agent resolves the target's domain name in dual-stack (IPv4/IPv6) environments. Defaults to Agent's policy.
SIP Server Test Additional Settings
Description (optional)
An optional, user-defined description for the test.
Alert suppression window
Temporarily disables all alerts for this test for a defined period.
SIP Server Test Outputs
SIP server tests enable access to the SIP server metrics, End-to-End Metrics, Path Visualization and BGP Route Visualization views.
RTP Stream Test
RTP Stream test measures the quality of Real-Time Protocol Voice stream between ThousandEyes agents acting as VoIP User Agents. An RTP Stream test establishes a point-to-point connection from the sender (VoIP call initiator) to the receiver (VoIP call endpoint). A stream of voice packets is sent between the endpoints to measure Mean Opinion Score (MOS), packet loss, discards, latency and Packet Delay Variation (PDV). The RTP Stream test also provides a choice in the Differentiated Services Code Point (DSCP) and codec values. For guidance on creating RTP Stream test, see Manually Configuring RTP Stream Tests.
Typical Use Cases
Identify the node responsible for degraded RTP Stream.
Analyze effects of BGP Route changes on RTP Stream.
Measure call audio quality in terms of
Mean Opinion Score
Loss
Discards
Latency
Packet Delay Variation
Example RTP Stream Test Results
Here is an RTP Stream test between two Cloud Agents displaying Packet Delay Variation
Other Included Tests
Agent-to-server test
BGP test
Manually Configuring RTP Stream Tests
To configure an RTP Stream test:
In the ThousandEyes platform, navigate to Network & App Synthetics > Test Settings.
Click the Add New Test button in the upper right corner of the screen.
From the menu that appears, select the card titled RTP Stream, located on the bottom right.
The configuration page is divided into the following sections: Basic Settings, Network and Voice Settings, and Additional Settings.
RTP Stream Test Basic Settings
Test name (optional)
A name for the test. This auto-populates with the target agent name but can be manually edited.
How often test runs
The frequency at which the test runs. Defaults to 2 minutes. More frequent intervals are recommended for critical services.
Where test runs from
Select one or more source agents to send the RTP stream from.
Where test runs to
Select a single target agent to receive the RTP stream.
Alerts
Enable or disable alerts for this test and assign one or more alert rules. Enabled by default; default alert rules are selected when available.
Tags
Assign tags to this test for filtering and organization. No tags are assigned by default.
RTP Stream Network and Voice Settings
Test execution schedule
Determines the timing of test rounds for all assigned agents.
Default: All agents start the test simultaneously.
Randomized Start Time: Staggers agent start times randomly within the test interval.
Server port
The UDP port on the target agent that will receive the RTP stream. Defaults to 49152. Change this to avoid conflicts or comply with firewall rules.
Codec
The voice codec to simulate. Defaults to G.711. This determines the bandwidth and quality characteristics of the RTP stream. Options range from high-fidelity (G.711) to low-bandwidth (G.729a).
Duration
The length of time the RTP stream is sent during each test round. Defaults to 5 seconds. Shorter durations are for quick checks, while longer durations (up to 30 seconds) can help capture transient issues.
De-jitter buffer size
The size of the simulated de-jitter buffer on the receiving agent. Defaults to 40 ms. A smaller buffer has less delay but is more sensitive to jitter, while a larger buffer can smooth out more jitter at the cost of increased delay.
Collect BGP data
Correlates network data with BGP routing information. Enabled by default.
Number of path traces
Sets the number of path traces performed in each test round to help discover multiple network paths. Defaults to 3. Custom values can range from 3 to 10.
DSCP
Sets the Differentiated Services Code Point (DSCP) value in the IP header of test packets to simulate a specific Quality of Service (QoS) policy. Defaults to EF (DSCP 46) for high-priority voice traffic.
RTP Stream Additional Settings
Description (optional)
An optional, user-defined description for the test.
Alert suppression window
Temporarily disables all alerts for this test for a defined period.
RTP Stream Test Outputs
RTP Stream tests enable access to the Voice Metrics, Path Visualization and BGP Route Visualization views.
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