Testing Sip Signaling Calls; Configuring Test Call Endpoints - AudioCodes Mediant 800 User Manual

CHAPTER 69 Testing SIP Signaling Calls
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Testing SIP Signaling Calls

A simulated endpoint can be configured on the device to test SIP signaling of calls between it and a
remote destination. This feature is useful in that it can remotely verify SIP message flow without
involving the remote end side in the debug process. The SIP test call simulates the SIP signaling
process - call setup, SIP 1xx responses, through to completing the SIP transaction with a 200 OK.
The test call sends Syslog messages to a Syslog server, showing the SIP message flow, tone
signals (e.g., DTMF), termination reasons, as well as voice quality statistics and thresholds (e.g.,
MOS).

Configuring Test Call Endpoints

The Test Call Rules table lets you test SIP signaling (setup and registration) and media (DTMF
signals) of calls between a simulated phone on the device and a remote IP endpoint. These tests
involve both incoming and outgoing calls, where the test endpoint can be configured as the caller or
called party. The simulated phone and remote endpoints are defined as SIP URIs (user@host) and
the remote endpoint can be defined as an IP Group or IP address.
Test calls can be dialed automatically at a user-defined interval and/or manually when required.
When a SIP test call is initiated, the device generates a SIP INVITE towards the remote endpoint
(e.g., a SIP proxy server or softswitch). It simulates the SIP call setup process, managing SIP 1xx
responses and completing the SIP transaction with a 200 OK.
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The following procedure describes how to configure test call rules through the Web interface. You
can also configure it through ini file [Test_Call] or CLI (configure troubleshoot > test-
call test-call-table).
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To configure a test call rule:
1. Open the Test Call Rules table (Troubleshoot menu > Troubleshoot tab > Test Call folder >
Test Call Rules).
2. Click New; the following dialog box appears:
By default, you can configure up to five test call rules. However, you can increase
this number by installing a License Key that licenses the required number. For more
information, contact the sales representative of your purchased device.
The device supports up to 400 concurrent test calls with PRT and up to 64
concurrent test calls with DTMF.
The device's Call Admission Control (CAC) feature (see
Admission Control on page 778) does not apply to Test Calls.
When the device operates in High-Availability (HA) mode, current Test Calls are
disconnected during an HA switchover.
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Configuring Call