Configuring Test Call Endpoints

The Test Call Rules table lets you test SIP signaling (setup and registration) and media (DTMF signaling) of calls between a simulated phone on the device and a remote IP endpoint. You can test incoming and outgoing calls, where the test endpoint can be configured as the caller or called party.

The device's simulated phone and the remote endpoints are defined by SIP URIs (user@host). The remote endpoint can be defined as an IP Group or IP address.

Test calls can be dialed automatically at a user-defined interval, or manually when required. When the device initiates a SIP test call, it sends a SIP INVITE towards the remote SIP User Agent (e.g., a SIP proxy server or softswitch). The device simulates the SIP call setup process, managing SIP 1xx responses and completing the SIP transaction with a SIP 200 OK after a user-defined duration.

When the remote SIP UA initiates the call with the device's test call endpoint, the test call ends when the remote UA ends the call (i.e., sends a SIP BYE message). Alternatively, the duration of the test call can be determined by the incoming SIP INVITE message, as described in Using SIP INVITE to Specify Test Call Duration.

● 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's Call Admission Control (CAC) feature (see Configuring Call Admission Control) doesn't apply to Test Calls.
● When the device operates in High-Availability (HA) mode, current Test Calls are disconnected during an HA switchover.
● To configure a Test Call endpoint for measuring and reporting MOS to WebRTC clients, see Reporting MOS Triggered by WebRTC Client
● You can also run a test call using the device's REST API. For more information, refer to the document REST API for SBC-Gateway-MSBR Device, by clicking here.
● For testing incoming calls, the device first tries to find a matching rule in the Test Calls Rules table (i.e., SIP INVITE Request-URI user part equals the 'Endpoint URI' parameter value). If there is no matching rule, the device then checks if the prefix of the user part matches the 'Test Call ID' parameter value configured for the Basic Test Call feature (see Configuring Basic Incoming Test Calls).

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).

➢ To configure a Test Call:
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:

3. Configure a test call according to the parameters described in the table below.
4. Click Apply, and then save your settings to flash memory.

Test Call Rules Table Parameter Descriptions

Parameter

Description
Common

'Index'

Defines an index number for the new table row.

Note: Each row must be configured with a unique index.

'Endpoint URI'

endpoint-uri

[Test_Call_EndpointURI]

Defines the endpoint's URI. This can be defined as a user or user@host. The device identifies this endpoint only by the URI's user part. The URI's host part is used in the SIP From header in REGISTER requests.

The parameter is used as follows:

■ Incoming test calls: The parameter defines the destination URI in the user part of the incoming INVITE Request-URI.
■ Outgoing test calls: The parameter defines the source URI in the user part of the outgoing INVITE request's From header.

The valid value is a string of up to 150 characters. By default, the parameter is not configured.

Note: The parameter is mandatory.

'Called URI'

called-uri

[Test_Call_CalledURI]

Defines the destination (called) URI (user@host) in the user part of the outgoing SIP INVITE Request-URI.

The valid value is a string of up to 150 characters. By default, the parameter is not configured.

Note: The parameter is applicable only to outgoing test calls.

'Route By'

route-by

[Test_Call_RouteBy]

Defines the type of routing method. This applies to incoming and outgoing calls.

■ [1] IP Group = (Default) Calls are matched by (or routed to) an IP Group. To specify the IP Group, see the 'IP Group' parameter in the table.
■ [2] Dest Address = Calls are matched by (or routed to) a destination IP address. To configure the address, see the 'Destination Address' parameter in the table.

Note:

■ If configured to Dest Address:
✔ You must assign a SIP Interface (see the 'SIP Interface' below).
✔ The IP Profile of the default IP Group (ID 0) is used. You can use a different IP Profile, by specifying an IP Group in the 'IP Group' parameter (below).
■ For REGISTER messages, if configured to IP Group, only Server-type IP Groups can be used.

'IP Group'

ip-group-id

[Test_Call_IPGroupName]

Assigns an IP Group where the test call is sent to or received from.

By default, no value is defined.

To configure IP Groups, see Configuring IP Groups.

Note:

■ The parameter is applicable if you configure the 'Route By' parameter to IP Group.
■ You can also use this parameter if you configure the 'Route By' parameter to Dest Address. This allows you to associate an IP Profile (that is assigned to the specified IP Group) with the Test Call. The Test Call is not routed to the IP Group, but uses only its IP Profile.
■ The IP Group is used for incoming and outgoing calls.

'Destination Address'

dst-address

[Test_Call_DestAddress]

Defines the destination host.

The valid value is an IP address[:port] in dotted-decimal notation or a DNS name[:port].

Note: The parameter is applicable only if you configure the 'Route By' parameter to Dest Address.

'SIP Interface'

sip-interface-name

[Test_Call_SIPInterfaceName]

Assigns a SIP Interface. This is the SIP Interface to which the test call is sent and received from.

By default, no value is defined.

To configure SIP Interfaces, see Configuring SIP Interfaces.

Note: The parameter is applicable only if the 'Route By' parameter is configured to Dest Address.

'Application Type'

application-type

[Test_Call_ApplicationType]

Defines the application type for the endpoint. This associates the IP Group and SRD to a specific SIP interface.

■ [2] SBC = SBC application

Note: The parameter must always be configured to SBC.

'Destination Transport Type'

dst-transport

[Test_Call_DestTransportType]

Defines the transport type for outgoing calls.

■ [-1] = Not configured (default)
■ [0] UDP
■ [1] TCP
■ [2] TLS
■ [3] SCTP

Note: The parameter is applicable only if you configure the 'Route By' parameter to Dest Address.

'QoE Profile'

qoe-profile

[Test_Call_QOEProfile]

Assigns a QoE Profile to the test call.

By default, no value is defined.

To configure QoE Profiles, see Configuring Quality of Experience Profiles.

'Bandwidth Profile'

bandwidth-profile

[Test_Call_BWProfile]

Assigns a Bandwidth Profile to the test call.

By default, no value is defined.

To configure Bandwidth Profiles, see Configuring Bandwidth Profiles.

Media

'Offered Audio Coders Group'

offered-audio-coders-group-name

[Test_Call_OfferedCodersGroupName]

Assigns a Coder Group, configured in the Coders Groups table, whose coders are added to the SDP Offer in the outgoing Test Call.

If not configured, the device uses the Coder Group specified by the 'Extension Coders Group' parameter of the IP Profile associated with the rule's IP Group (see the 'IP Group' parameter above).

To configure Coder Groups, see Configuring Coder Groups.

Note:

■ The parameter's settings override the corresponding parameter of the IP Profile associated with the rule's IP Group.
■ If you don't configure this parameter nor the corresponding parameter of the associated IP Profile, the device uses Coder Group ID 0.

'Allowed Audio Coders Group'

allowed-audio-coders-group-name

[Test_Call_AllowedAudioCodersGroupName]

Assigns an Allowed Audio Coders Group, configured in the Allowed Audio Coders Groups table, which defines only the coders that can be used for the test call. For incoming test calls, the device accepts the first offered coder that is supported and allowed.

If not configured, the device uses the Allowed Audio Coders Group specified by the 'Allowed Audio Coders' parameter of the IP Profile associated with the rule's IP Group (see the 'IP Group' parameter above).

To configure Allowed Audio Coders Groups, see Configuring Allowed Audio Coder Groups.

Note: The parameter's settings override the corresponding parameter of the IP Profile associated with the rule's IP Group.

'Allowed Coders Mode'

allowed-coders-mode

[Test_Call_AllowedCodersMode]

Defines the mode of the Allowed Coders feature for the Test Call.

■ [-1] Not Configured = (Default) The mode is according to the 'Allowed Coders Mode' parameter of the IP Profile associated with the rule's IP Group (see the 'IP Group' parameter above).
■ [0] Restriction = The device uses only allowed coders as configured in the 'Allowed Audio Coders Group' parameter (above) and removes all other coders from the SDP offer. If you have also configured additional coders in the 'Offered Audio Coders Group' parameter (above), then these coders are added to the SDP offer if they appear in the assigned Allowed Audio Coders Group.
■ [1] Preference = The device re-arranges the priority (order) of the coders in the incoming SDP offer according to their order of appearance in the Allowed Audio Coders Group. The coders in the original SDP offer are listed after the allowed coders.
■ [2] Restriction and Preference = The device uses both the Restriction and Preference options.

Note: Except for Not Configured, the parameter's settings override the corresponding parameter of the IP Profile associated with the rule's IP Group.

'Media Security Mode'

media-security-mode

[Test_Call_MediaSecurityMode]

Defines the handling of RTP and SRTP.

■ [-1] Not Configured = (Default) Handling is according to the 'SBC Media Security Mode' parameter of the IP Profile associated with the rule's IP Group (see the 'IP Group' parameter above).
■ [0] As is = No special handling for RTP\SRTP is done.
■ [1] SRTP = Only SRTP media lines are negotiated and RTP media lines are removed from the incoming SDP offer-answer.
■ [2] RTP = Only RTP media lines are negotiated and SRTP media lines are removed from the incoming SDP offer-answer.
■ [3] Both = Each SDP offer-answer is extended (if not already) to two media lines - one for RTP and one for SRTP.

Note:

■ To enable SRTP, configure the [EnableMediaSecurity] parameter to [1].
■ Except for Not Configured, the parameter's settings override the corresponding parameter of the IP Profile that is associated with the rule's IP Group.

'Play DTMF Method'

play-dtmf-method

[Test_Call_PlayDTMFMethod]

Defines the method used by the devicefor sending DTMF digits that are played to the called party when the call is answered.

■ [-1] Not Configured = The mode is according to the 'Alternative DTMF Method' and 'RFC 2833 Mode' parameters of the IP Profile associated with the rule's IP Group (see the 'IP Group' parameter above).
■ [0] RFC 2833 = (Default) The device sends the DTMF digits using the RFC 2833 method (out-of-band).
■ [1] In Band = The device sends the DTMF digits in-band (in the RTP stream).

Note:

■ The parameter is applicable only if you configure the 'Play' parameter to DTMF.
■ Playing DTMF digits requires DSP resources when the DTMF method is In Band.
■ If the Test Call sends the SDP offer, the recommended DTMF configuration of the associated IP Profile is as follows:
✔ For RFC 2833: 'RFC 2833 Mode' = Extend; 'Alternative DTMF Method' = As Is
✔ For In Band: 'RFC 2833 Mode' = Disallow; 'Alternative DTMF Method' = As Is
■ If the Test Call receives the SDP offer, the recommended configuration is as follows (i.e., incoming SDP offer determines the method): 'RFC 2833 Mode' = As Is; 'Alternative DTMF Method' = As Is I

Authentication

Note: These parameters are applicable only if the 'Call Party' parameter (below) is configured to Caller.

'Auto Register'

auto-register

[Test_Call_AutoRegister]

Enables automatic registration of the endpoint. The endpoint can register to the device itself or to the 'Destination Address' or 'IP Group' parameter settings (see above).

■ [0] Disable (default)
■ [1] Enable

'Username'

user-name

[Test_Call_UserName]

Defines the authentication username.

The valid value is a string of up to 60 characters. By default, no value is defined.

'Password'

password

[Test_Call_Password]

Defines the authentication password.

By default, no password is defined.

Note: The parameter cannot be configured with wide characters.

Test Setting

'Call Party'

call-party

[Test_Call_CallParty]

Defines if the test endpoint is the initiator (caller) or receiving side (called) of the test call.

■ [0] Caller = (Default) The device's test endpoint is the calling party (i.e., applicable only to outgoing test calls).
■ [1] Called = The device's test endpoint is the called (callee) party (i.e., applicable only to incoming test calls).

'Maximum Channels for Session'

max-channels

[Test_Call_MaxChannels]

Defines the maximum number of concurrent channels for the test session. For example, if you have configured an endpoint "101" and you configure the parameter to "3", the device automatically creates three simulated endpoints - "101", "102" and "103" (i.e., consecutive endpoint URIs are assigned).

The default is 1.

'Call Duration'

call-duration

[Test_Call_CallDuration]

Defines the call duration (in seconds).

The valid value is -1 to 100000. The default is 20. A value of 0 means infinite. A value of -1 means that the parameter value is automatically calculated according to the values of the 'Calls per Second' and 'Maximum Channels for Session' parameters.

Note: The parameter is applicable only if you configure 'Call Party' to Caller.

'Calls per Second'

calls-per-second

[Test_Call_CallsPerSecond]

Defines the number of calls per second.

Note: The parameter is applicable only if you configure 'Call Party' to Caller.

'Test Mode'

test-mode

[Test_Call_TestMode]

Defines the test session mode.

■ [0] Once = (Default) The test runs until the lowest value between the following is reached:
✔ Maximum channels is reached for the test session, configured by 'Maximum Channels for Session'.
✔ Call duration ('Call Duration') multiplied by calls per second ('Calls per Second').
✔ Test duration expires, configured by 'Test Duration'.
■ [1] Continuous = The test runs until the configured test duration is reached. If it reaches the maximum channels configured for the test session (in the 'Maximum Channels for Session'), it waits until the configured call duration of a currently established tested call expires before making the next test call. In this way, the test session stays within the configured maximum channels.

Note: The parameter is applicable only if you configure 'Call Party' to Caller.

'Test Duration'

test-duration

[Test_Call_TestDuration]

Defines the test duration (in minutes).

The valid value is 0 to 100000. The default is 0 (i.e., unlimited).

Note: The parameter is applicable only if you configure 'Call Party' to Caller.

'Play'

play

[Test_Call_Play]

Enables the playing of a tone to the answered side of the call.

■ [0] Disable = No tone is played.
■ [1] DTMF = (Default) Plays (loop) a user-defined DTMF string, which is configured in Configuring DTMF Tones for Test Calls.
■ [2] PRT = Plays (loop) a pre-recorded tone (audio file) from the PRT file that is installed on the device. You can either specify the tone (by index) to play from the PRT file in the 'Play Tone Index' parameter (below), or implement a basic NetAnn feature whereby the tone from the PRT file (and other characteristics) are specified by NetAnn parameters in the Request-URI of the incoming SIP INVITE message. When using NetAnn, instead of connecting the call (i.e., 200 OK), the devicereplies with a SIP 183 containing SDP.

The NetAnn parameters include the following:

● early=yes: Indicates that NetAnn is used for playing the tone.
● play=<Prompt/Tone Index in PRT file>: Defines the tone to play from the PRT file.
● repeat=<Times>: Defines how many times the tone is played (loops) before the device disconnects the call.
● delay=<Delay Time>: Defines the delay time (in msec) between each played (loop) tone. If the parameter is not present, the default is 2,000 ms (2 seconds).

The following shows an example of a Request-URI with NetAnn parameters that instruct the device to play three times (loops) the tone that is defined at Index 15 in the PRT file: 

INVITE sip:200@1.1.1.1;early=yes;play=15;repeat=3

Note:

■ You can configure the DTMF signaling type (RFC 2833 or in-band), using the 'Play DTMF Method' parameter (above).
■ Playing a tone from the PRT file requires DSP resources if the coder with which the tone was created is different to the coder used for the Test Call.
■ You can also use NetAnn parameters to play a specific recorded tone to the caller (source) when the destination fails for a regular IP-to-IP SBC call. To configure this:
✔ Configure a Message Manipulation rule that adds the NetAnn parameters, based on the tone that you want played, to the INVITE message's Request-URI.
✔ Configure a Number Manipulation rule that changes the destination number to the Test Call ID.
✔ Configure an IP Group to represent the device itself (which will be the Test Call module) and assign it the Message Manipulation rule and the Number Manipulation rule.
✔ Configure an alternative routing rule in the IP-to-IP Routing table that re-routes the call to the IP Group presenting the Test Call module.

When the IP-to-IP call fails, the device uses the alternative routing rule to re-route the call to the Test Call module, which sends a SIP 183 response to the caller, playing the specified tone.

'Play Tone Index'

play-tone-index

[Test_Call_PlayToneIndex]

Defines the tone that you want played from the installed PRT file, to the called party when the call is answered.

The valid value is the index number (1-80) of the tone in the PRT file. By default (-1), the device plays the tone defined at index 22 "acDialTone2".

Note:

■ The parameter is applicable only if you configure the 'Play' parameter to PRT.
■ To play user-defined tones, you need to record your tones and then install them on the device using a loadable Prerecorded Tones (PRT) file, which is created using AudioCodes DConvert utility. When you create the PRT file, each recorded tone file must be added to the PRT file with the tone type "acUserDefineTone<Index>". For more information, see Prerecorded Tones File.

'Schedule Interval'

schedule-interval

[Test_Call_ScheduleInterval]

Defines the interval (in minutes) between automatic outgoing test calls.

The valid value range is 0 to 100000. The default is 0 (i.e., scheduling is disabled).

Note: The parameter is applicable only if you configure 'Call Party' to Caller.