Voice Tests
Last updated
Last updated
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.
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.
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.
Here is a SIP Server test targeting an internal VoIP server.
Agent-to-server test
SIP Server: Specify domain name or IP address of the SIP server.
SIP Proxy: Checkbox to enable the use of SIP Proxy, and text field to enter Proxy's IP address.
Protocol: Select either TCP or UDP.
Port: TCP or UDP port number on which the SIP service is listening.
Perform SIP Register: Enable/disable SIP registration as part of the test.
User: Username for SIP registration.
Auth User: Alternative username when performing authentication (the authuser setting).
Password: Password or secret for authentication.
Desired status code: Sets the SIP status code returned by the server which will be considered a successful test (no errors will be displayed in the test table results, no response code-based alerts generated, etc...). By default, either a 200-level or 300-level status code is considered a successful status code. Uncheck the box and enter a code in the field below to use a different status code.
Verify Headers: Search SIP response headers for text which matches the expression in the Verify Headers field. The expression can be literal text or a POSIX regular expression. If no match occurs, the test shows an error.
SIP server tests enable access to the SIP server metrics, End-to-End Metrics, Path Visualization and BGP Route Visualization views.
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 RTP Stream Test Settings.
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
Here is an RTP Stream test between two Cloud Agents displaying Packet Delay Variation
Agent-to-server test
BGP test
Test type: RTP Stream.
Test Name: __This optional parameter gives the test a label. When no label is provided, the the values in the Target and the Server Port field will be combined to comprise the test name. A test name cannot exceed 255 characters.
Target: Enterprise Agent or Cloud Agent.
Interval: How frequently the test will be run.
Agents: This drop-down lists the ThousandEyes Cloud Agents and (optionally) Enterprise Agents agents that are available to your account. Select one or more Agents to assign them to this test.
Alerts: When the Enable box is checked, the Alert Rules selected in drop-down list will be active for the test. You can select Alert Rules with the drop-down list, and create, modify and delete Alert Rules with the Edit Alert Rules link.
Server Port: The port number that the server uses for incoming RTP sessions.
Codec: The codec name (and associated bit rate) used for the RTP session.
Duration: The length of time the test will run, in seconds.
De-jitter Buffer Size: The size of the de-jitter buffer, in seconds. De-jitter buffers store traffic in order to eliminate the effects of delay variations. Having too small or too large a buffer size can result in audio gaps perceived by the call recipient.
Collect BGP data: Check this box to enable the BGP Path Visualization view. The BGP Public Monitors option button allows you to choose whether ThousandEyes' public BGP Monitors should be used for monitoring target prefixes. The Private BGP Monitors drop-down box allows you to select which private BGP Monitors should be used for monitoring target prefixes. By default, all public and private monitors are selected.
No. of Path Traces: Three path trace packets are used by each Agent by default to discover each hop in the Path Visualization to the target. Uncheck the box to display a slider which allows selection of 1 - 10 packets.
DSCP: The Differentiated Services Code Point (DSCP) value for the IP packet headers of the RTP session. DSCP allows for prioritization of voice traffic in a network.
RTP Stream tests enable access to the Voice Metrics, Path Visualization and BGP Route Visualization views.
