> For the complete documentation index, see [llms.txt](https://docs.thousandeyes.com/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://docs.thousandeyes.com/product-documentation/tests/voice-tests.md).
# 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.
## 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](https://docs.thousandeyes.com/product-documentation/internet-and-wan-monitoring/tests/voice-tests/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:
1. In the ThousandEyes platform, navigate to **Network & App Synthetics > Test Settings**.
2. Click the **Add New Test** button in the upper right corner of the screen.
3. From the menu that appears, select the card titled **SIP Server**, located on the bottom left.
4. The configuration page is divided into the following sections: Basic Settings, Server Configuration and Network Settings, and Additional Settings.
### SIP Server Test Basic Settings
| Setting | Description |
| ------------------------ | ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **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
| Setting | Description |
| ---------------------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ |
| **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
| Setting | Description |
| ---------------------------- | ------------------------------------------------------------------- |
| **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](#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:
1. In the ThousandEyes platform, navigate to **Network & App Synthetics > Test Settings**.
2. Click the **Add New Test** button in the upper right corner of the screen.
3. From the menu that appears, select the card titled **RTP Stream**, located on the bottom right.
4. The configuration page is divided into the following sections: Basic Settings, Network and Voice Settings, and Additional Settings.
#### RTP Stream Test Basic Settings
| Setting | Description |
| ------------------------ | ----------------------------------------------------------------------------------------------------------------------------------------------- |
| **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
| Setting | Description |
| --------------------------- | ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **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
| Setting | Description |
| ---------------------------- | ------------------------------------------------------------------- |
| **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.