SIP Actions
In VoLTE testing, the SIP VoIP Supplementary option, when enabled, provides users with a way for creating SIP tests using scripts and actions. The main advantage of using scripts and actions is to allow users to create different SIP message flows tailored to their particular test environments. Additional details in the SIP Editor.
Landslide provides the following list of actions. Some actions may have multiple pre-configured options and they are interpreted as follows:
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Success – an action is expected to send/receive a success response code (i.e. 200-OK).
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Fail – an action is expected to send/receive a failure response code (i.e. 480-Busy)
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No auth – an action doesn’t expect to be authenticated (challenged by proxy/registrar).
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Auth req – an action is expected to be authenticated and will produce appropriate responses when challenged.
NOTE: If your saved tests contain an older version of a SIP Script Built-in Action, the TAS will automatically upgrade it to match the properties of the current version. The Caller Assignment and Action Roles will be refreshed and you may see Upgrade Warnings such as this: - PRESS F1 FOR REPAIR/UPGRADE HELP
- Gm SIP - SIP Script [SipFlow] SIP Script [SipFlow] Action(1) Caller Assignments Upgraded
- Gm SIP - SIP Script [SipFlow] SIP Script [SipFlow] Action(3) Caller Assignments Upgraded
- Gm SIP - SIP Script [SipFlow] SIP Script [SipFlow] Action(5) Caller Assignments Upgraded
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The validation rules have changed with regards to how we keep the configured version of DTMF/Tones versus the Action way of DTMF/Tones mutually exclusive. Only the "Media_SendWait_DTMFString" Action will be prevented when RFC2833/RFC4733 configuration is enabled. Up until now any Action with "SendWait" in the title would be applied to this rule. We have added additional validation of the MEDIA_CHOICE aka "Media" property and the RTP_VOICE_DMF aka "RTP Voice DMF" property. The validation will make sure that DMFs are configured where they should be. For MEDIA_CHOICE, if the user chooses a specific DMF, the validation will make sure a DMF is configured at that index. For RTP_VOICE_DMF, if the user chooses ALL, the validation will make sure there is at least one rtpvoice DMF. If the user chooses a specific DMF, the validation will make sure that there is an rtpvoice DMF at that index. |
| SIP Built In Actions: Enabled from SIP VoIP Supplementary Option. | ||
| Call_Connect (success, no auth), Call_Connect (Success, Auth req) Call_connect (fail, no auth) |
The “Call_Connect” action is used for connecting 2 callers within a script. In this action, a caller, whose role is the source, will formulate and send an SIP INVITE request to a caller whose role is the destination. Also, users can select one of the following media options for specifying which media will be used for connecting:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_OnHold (success,no auth), Call_OnHold (fail, no auth) |
The “Call_OnHold” action is used for placing 2 connected callers on-hold. In this action, a caller, whose role is the source, will formulate and send an SIP re-INVITE request to a caller whose role is the destination. Users can select one of the following media options for specifying which active media will be put on-hold:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.  
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| Call_OffHold (success, no auth), Call_OffHold (fail, no auth) |
The “Call_OffHold” action is used for resume media of 2 connected callers. In this action, a caller, whose role is the source, will formulate and send an SIP re-INVITE request to a caller whose role is the destination. Users can select one of the following media options for specifying which on-hold media will be resumed:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_Disconnect (success, no auth) | The “Call_Disconnect” action is used for disconnecting 2 callers within a script. In this action, a caller, whose role is the source, will formulate and send an SIP BYE request to a caller whose role is the destination.
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. Note: BYE will be sent for a call which is in an established state as well as in a Proceeding State. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_ForwardUnconditional (success, no auth) Call_ForwardUnconditional (fail, no auth) | The “Call_ForwardUnconditional” action is used for testing forwarding function of a proxy. (NOTE: it requires that proxy has supports for forwarding function and configured properly for this action to work.) In this action, a caller whose role is the source will formulate and send SIP INVITE request to a caller whose role is original-destination (see illustration below) but it is expected a caller whose role is the destination to response. Users can select one of the following media options for specifying which media will be used for connecting:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_ForwardBusy (success, no auth) Call_ForwardBusy (fail, no auth) | The “Call_ForwardBusy” action is used for testing forwarding function of a proxy (NOTE: it requires that proxy has supports for forwarding function and configured properly for this action to work.) If Bob wants calls to B forwarded to C if B is busy. User A calls B, B is busy, the proxy server places call to C. Upon success, RTP will be established. Users can select one of the following media options for specifying which media will be used for connecting:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_Refer (success, no auth), Call_Refer (fail, no auth), | The “Call_Refer” action is used for referring one caller to connect to another caller. In this action, the caller whose role is the source will formulate and send SIP REFER request to a caller whose role is the destination. The SIP REFER request will contain address information of the caller whose role is the REFER (see illustration below).
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked. Refer Method - Options : INVITE, REFER, BYE (asking a user to join a conference, asking a conference-server to invite a user, asking a conference-server to disconnect a user).
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| Call_ConnectRefer (success, no auth), Call_ConnectRefer (fail, no auth) | The “Call_ConnectRefer” action is used for connecting 2 callers similar to “Call_Connect” action except it use previously received referred information in constructing the SIP INVITE request message. NOTE: this action should be used after a Call_Refer action.
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_Notify (Success, no auth) Call_Notify (fail, no auth) | The “Call_Notify” action is used for sending SIP NOTIFY request from one caller to another. Use the source and destination option for specifying which caller will be a sender of a SIP NOTIFY message. This action must be used after a subscription between 2 callers has been established. A subscription can be established by using the Call_Refer action.
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.  
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| Call_Subscribe (success, no auth), Call_Subscribe (Success, Auth req), Call_Subscribe (fail, no auth) |
Call_Subscribe. Select for In-Call Subscribe for an event or a state of a resource with authentication. Call_Subscribe (success, no auth) – Purpose: This action is used for establishing an in-call subscription between a selected source and destination. This action does not expect a challenge-response from a receiver. Prerequisite: There must exist one established/ongoing call between a source and destination. Action will fail if there is no existing between a source and destination. Use the optional Call ID Number entry to specific a specific call if there're more ongoing calls between a source and destination. Call_Subscribe (success, w/ auth) – similar to the above Call_Subscribe (success, no auth) except this action expects to receive an authentication-challenge response from remote entity. Call_Subscribe (fail, no auth) – used for initiating a SIP SUBSCRIBE request within an existing call. This action expects to receive a failure response from remote entity. Parameters:
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| Call_Notify (success, no auth), Call_Notify (Success, Auth req), Call_Notify (fail, no auth) |
Call_Notify (success, no auth) . In-Call Notify subscribers about an event or a state of a resource without authentication. Used for initiating a SIP NOTIFY request message containing a bare minimum required SIP header. Content of a SIP NOTIFY message can be changed via SIP message editor for simulating different. Call_Notify (success, no auth) – similar to the above Call_Notify(success, no auth) except this action expects to receive an authentication-challenge response from remote entity. Call_Notify (fail, no auth) - similar to the above Call_Notify(success, no auth) except this action expects to receive a failure response from remote entity. Parameters:
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| Call_Unsubscribe (success, no auth), Call_Unsubscribe (Success, Auth req), Call_Unsubscribe (fail, no auth) |
Call_Unsubscribe (success, no auth) – In-Call Unscubscribe a subscription without authentication. Used for initiating a SIP SUBSCRIBE request containing zero-value expiration within an existing call. Call_Unsubscribe (success, no auth) – similar to the above Call_Unsubscribe (success, no auth), except this action expects to receive an authentication-challenge response from remote entity. Call_Unsubscribe (fail, no auth) - similar to the above Unsubscribe (success, no auth) except this action expects to receive a failure response from remote entity. Parameters:
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| Call_UpdateRequest (Success, no auth) Call_UpdateRequest (fail, no auth) |
Call_Update Request.
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Call_ForwardNoAnswer (Success, no auth) Call_ForwardNoAnswer (fail, no auth) |
The “Call_ForwardNoAnswer” action is used for testing forwarding function of a proxy (NOTE: it requires that proxy has supports for forwarding function and configured properly for this action to work.) If Bob wants calls to B forwarded to C if B does not answer. User A calls B, B does not answer, the proxy server places call to C. Upon success, RTP will be established. Users can select one of the following media options for specifying which media will be used for connecting:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.
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| Wait_For_Time |
The “Wait_For_Time” action is used for telling one or more callers within a script to wait for (do not proceed to a next action within a script) a specified period in milliseconds. User can choose either ALL callers or for individual caller shown below. Select Sync to allow multiply parties to sync before executing the next script. With sync enabled, a party will wait until (whichever come first): This action is often used for synchronizing callers or for letting callers to continue sending data (RTP/MSRP) for a period of time before moving on to a next action. |
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| Media_Start | The “Media_Start” action is used for telling 2 callers to start sending media. Users can select one of the following media options for specifying which media will be started for an existing connection:
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| Media_Stop | The “Media_Stop” action is used for telling 2 callers to stop sending media. Users can select one of the following media options for specifying which media will be stopped for an existing connection:
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Media_SendWait_Silence
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Media_SendWait_Tones
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| Media_SendWait_DTMF_String |
DTMF tone detection and generation. This built-in action creates in-band DTMF tones for use with SUTs which use DTMF control such as conference bridges. It can be configured to send tones simulating conference bridge numbers and access codes. Wait Times and Media-Start/Stop can be used to simulate interaction and conversation with the SUT.
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Media_Wait_Voice |
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| Media_SendWait_Voice |
Added support for Voice Analysis - IVR - See Media QoS Settings Topic.
The action has to two parties: WAV file sender (Send Party) and receiver (Wait Party). Wait Party can wait for voice up to "Wait Time" before media traffic begins to come. Media can be a specific DMF or ALL. If ALL is specified at least one DMF should be of rtpvoice type. If a specific one (DMF2, in fact) is configured then it has to be rtpvoice DMF.
Limitation / Caveat:
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| Call_ReInvite (success , no auth) | Add Call_ReInvite action to re-negotiate media settings.
Media parameter values = ALL, or specific DMF (DMF1, DMF2, and etc.). The default is ALL.
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| Provisional (Unreliable), Provisional (reliable) | The “Provisional (unreliable)”, “Provisional (reliable)” action are used for instructing a callee (person who receives calls) to formulate and send SIP 1xx responses to caller (person who initiate calls). These provisional actions must be inserted into “Call_Connect” or “Call_ConnectRefer” actions because they are optional for a call establishment. They are useful for indicating call progress, ringing, status, etc. They can also be inserted in the following actions Pre_conditional Provisionals (Class 8) : Call_Connect, Call_Refer, Call_AddMedia, Call_RemoveMedia, Call_ForwardUnconditional, Call_ForwardNoAnswer, Call_ForwardBusy, Call_Waiting. In addition, they can be inserted in the following Provisional actions: Call_OnHold and Call_OffHold.  |
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| Subscribe (Success, no auth) Subscribe (fail, no auth) |
Subscribe for an event or a state of a resource with or without authentication. To configure landslide subscribers for subscribing to a status of a conference, set the 'Event' of the 'Subscribe' action to 'conference'. The event='conference' identifies the subscription is for status of a conference (RFC 3265, RFC 4575) and landslide subscriber will attempt to use a previously saved conference-id between a specified source and destination parties as a remote target of the subscription; if there is no saved conference-id, it will use configured information of a specified destination party in the phonebook as a remote target of the subscription. IMS-Node conference server does support this Subscription/Notification feature.
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| Notify (Success, no auth), Notify (Success, auth req), Notify (fail, no auth) |
Notify subscribers of an event or a state of a resource with or without authentication. The following limitation relates to NOTIFY messages.
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| Unsolicited_Notify (success, no auth) |
Unsolicited Notify about an event or a state of a resource without authentication.
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| Options (no auth), Options (no auth, fail) |
The SIP method OPTIONS allows a UA to query another UA or a proxy server as to its capabilities. This allows a client to discover information about the supported methods, content types, extensions, codes, etc. without "ringing" the other party. Refer to RFC 3261 (Section 11) for additional details. Perform an OPTIONS action without auth. Perform a failed OPTIONS action without auth.
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| Unsubscribe (Success, no auth), Unsubscribe (Success, auth req), Unsubscribe (fail, no auth) |
Unsubscribe a subscription with or without authentication.
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| Registration (success, no auth), Registration (success, auth req) |
The “Registration” action is used for instructing callers of a script to register with a configured registrar. The “no auth” option indicates registration without authentication; whereas, the “auth req” indicates registration with authentication.
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| Deregistration (success, no auth), Deregistration (success, auth req) | Deregistration (success, no auth), Deregistration (success, authreq)   |
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| Call_AddMedia (Success, no auth) Call_AddMedia (fail, no auth) | The “Call_AddMedia” action is used for adding additional media a previously established session between 2 callers. In this action, a caller whose role is the source will formulate and send SIP re-INVITE request to a caller whose role is original-destination. Users can select one of the following media options for specifying which media will be added for an existing connection:
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| Call_RemoveMedia (Success, no auth) Call_RemoveMedia (fail, no auth) | The “Call_RemoveMedia” action is used for removing existing media previously established between 2 callers. In this action, a caller whose role is the source will formulate and send SIP re-INVITE request to a caller whose role is original-destination. Users can select one of the following media options for specifying which media will be removed from an existing connection:
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| Call_CommunicationBarring (Success, no auth) |
Call Communications action involves 2 participants (A, B) and an application-server (AS). It demonstrates AS that rejects an outgoing call from A if the media type in SDP offer matches configured media type. 1) Call_CommunicationBarring (success, no auth) (A, B)
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| Group_Chat_Setup_5_Users | Setup Group chat for 5 Users. Select the Roles. INVITEe message will be sent from Source A to Conference Server (F). The INVITE message includes XML containing all the participants in the group chat. In the example below, Source A will include all the Destinations from B to E (B, C, D, E). Conference Server F on receiving an INVITE from A, will send 200 OK to A and will also generate individual INVITE towards each Participant. After this action, the Control path will be established.   |
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| Group_Chat_DataPathSetup5Users | Setup Group Chat Data Path (i.e. MSRP Sockets are in ready state). | |
| Group_Chat_MediaStart | Start Media. MSRP data is sent between Source/Destination to Conference server. On receiving Data, the Conference Server will Broadcast the data to it's participants. | |
| Group_Chat_DataPathTerminate5Users | Terminate Group chat data path for 5 users. (i.e. The MSRP data path is torn down). | |
| Publish (no auth), Publish (no auth, fail) |
SIP PUBLISH method is used for the publication of event state from a user agent to an entity that is responsible for composing this event state and distributing it to interested partied through the SIP Events framework. The first application of this extension is for the publication of presence information. Refer to chapter 4 of IMS book [6]. Perform a PUBLISH action without auth. Perform a failed PUBLISH action without auth. XML editor functionality is supported for SIP PUBLISH message.
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| Diversion_Service_Control (success) | Use for Star Dialing. Use for Activation or De-Activation of the service. On IMS Node Emulator Configuration - Application Servers - Enable built-in Application Servers - Communication Diversion. The App servers can be used to set up diversion type and control.   |
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Call_Waiting (success, no auth) Call_Waiting (fail, no auth) |
In call waiting, user B is in an active call with another user and then receives a call from user A. User B can place the first call on hold with the other user, and talk with caller A. User B can switch back and forth between callers. Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.    |
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| Group_Chat_SendText | Action to send a text/plain message - Part of Group Chat File Transferring feature.  |
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| Group_Chat_Send_CpimMessage | Action to send a Cpim Formatted message - Part of Group Chat File Transferring feature.  |
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| Group_Chat_Send_CpimFile | Action to send a file using Cpim Formatted message - Part of Group Chat File Transferring feature.  Added support for up to 10 TDF files:  |
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| Group_Chat_Send_CpimFileInfo | Action to send file information using Cpim Formatted message - Part of Group Chat File Transferring feature.  |
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| Group_Chat_HttpUploadFile | Action to send a file to an HTTP file server - Part of Group Chat File Transferring feature. Click here to learn how to configure DMF to perform HTTP Digest Authentication, Upload the file and extract the XML from Content Server reply.  |
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| Group_Chat_HttpDnloadFile | Action to retrieve a file to an HTTP file server - Part of Group Chat File Transferring feature. Click here to learn how to configure DMF to perform HTTP Digest Authentication and download the file from Content Server reply.  |
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Note: removed Precondition restriction |
Action to perform an Update. As of release 20.0, this action is now available in Supplementary script irrespective of Enable Precondition checkbox in the VoLTE tab. The UPDATE action is implemented only for "Call_Connect", "Call_OnHold", "Call_OffHold" , "Call_ForwardUnconditional", "Call_ForwardNoAnswer" "Call_ForwardBusy". While it can be added to other actions, it is not fully supported unless it is used in one of the actions listed above. When you select this action it will be inserted into your script :
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| MCPTT_Registration (success, no auth) | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select for MCPTT Registration without authentication. |
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| MCPTT_Registration (success, auth req) |
Part of the Mission Critical Push-To-Talk (MCPTT) services. Select for MCPTT Registration with authentication. |
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| MCPTT_Connect (success, no auth) | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to establish a MCPTT Connection (success, no auth).   |
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| MCPTT_Connect (Fail, no auth) | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to establish a MCPTT Connection (Fail, no auth).   |
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| MCPTT_PrivateCall_Connect (Success, no auth) | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to establish a MCPTT Private Call Connection.   |
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| MCPTT_Group_Configuration_Subscribe (success, no auth) | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select for an MCPTT Subscription to a Management Server for an event or a state of a resource without authentication.   |
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| MCPTT_GroupCallSetup5Users | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to establish a MCPTT Group Call with 5 Users.   |
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| MCPTT_GroupCall_Connect | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to establish a MCPTT Group Call with 5 Users. Used by the MCPTT Floor Control feature.   |
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| MCPTT_PrivateCall_Floor_Request | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to send a Floor Request for a MCPTT Private Call. Used by the MCPTT Floor Control feature.   |
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| MCPTT_PrivateCall_Floor_Release | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to send a Floor Release for a MCPTT Private Call. Used by the MCPTT Floor Control feature.   |
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| MCPTT_GroupCall_Floor_Request | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to send a Floor Request for a MCPTT Group Call. Used by the MCPTT Floor Control feature.   |
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| MCPTT_GroupCall_Floor_Release | Part of the Mission Critical Push-To-Talk (MCPTT) services. Select to send a Floor Release for a MCPTT Group Call. Used by the MCPTT Floor Control feature.   |
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| MCPTT_Message (no auth, success) | Select to perform an MCPTT MESSAGE Action without authentication. Select the XML Body Type from the list below: MBMS bearer announcement, MBMS listening status report, Location reporting configuration, Location information report  
3 new Body Parts that can be added to SIP_REQ_MCPTT_MESSAGE :
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| VMS_Connect (Success, no auth) | Connect to a Virtual Voice Message Server (vVMS). All SIP INVITE Requests will use a multipart body (SDP and Geolocation).
 
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| VMS_Disconnect ( no auth) | Disconnect the call from a VMS Server. All SIP INVITE Requests will use a multipart body (SDP and Geolocation).   |
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| Media_DtmfStart | Start Dtmf. Enable RTP RFC2833/RFC4733 with Media_DtmfStart. Enter DTMF String. Range: 1 to 32 DTMF characters (Valid Characters are 0 – 9, *, #, A – D) Default: 12345
Enter DTMF Duration (ms). DTMF Duration can be configured as a single value that will be applied to all digits in DTMF String. Prior to 19.8 each digit in DTMF String required a correspondent value one Duration specified.
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| Media_DtmfStop | Stop Dtmf.   |
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| VMS_CallBackWithINVITE (success, no auth) | INVITE Call Back from VMS Server. All SIP INVITE Requests will use a multipart body (SDP and Geolocation).  |
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| VMS_CallbackWithREFER ( no auth) | REFER Call Back from VMS Server. All SIP INVITE Requests will use a multipart body (SDP and Geolocation).   |
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| Disconnect with ACR (no auth) | Disconnect from VMS Server with Anonymous Call Rejection (ACR)   |
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| Disconnect for INVITE callback | Disconnect for INVITE callback received from VMS. All SIP INVITE Requests will use a multipart body (SDP and Geolocation).   |
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| Call_610 ( success, no auth) | Call to 610. All SIP INVITE Requests will use a multipart body (SDP and Geolocation).   |
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| Media_Pause | Media pause.   |
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| Media_Resume | Media resume.   |
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| Call_Attempt (with auth, no auth) |
Call attempt with or without authorization. Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked. Call_Attempt (no auth) action has been expanded to include an option to handle Fork Call - Call Leg Forked. Options : No-Fork, Handle-Fork-At-Source. When Handle-Fork-At-Source is selected:
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| Call_Accept | Call Accept. Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.   |
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| Call_Reject | Call Reject. Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.   |
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| Call_Attempt_Three_Party |
Call Attempt Three Party (No Auth) action. 1. This action is typically used when User A is attempting to call user B, however B does not answer and then call is forwarded to some other entity such as voicemail. A Typical call flow looks as follow:
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Part of the Mission Critical Push-To-Talk (MCPTT) services. Used to trigger Temporary Mobile Group Identity TMGI (s) Allocation / Deallocation. Enter Number of Allocations. Available when MB2 interface is enabled.
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MBMS_Bearer_Control |
Part of the Mission Critical Push-To-Talk (MCPTT) services. Used to trigger Activate / Deactivate / Modify MBMS Bearer Procedure. Added the option to add TMGI Index (es). Available when MB2 interface is enabled.
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MCCP_Map_Group_to_Bearer |
Part of the Mission Critical Push-To-Talk (MCPTT) services. Used to trigger MCCP Map Group to Bearer Procedure. Available when MB2 interface is enabled.
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MCCP_Unmap_Group_to_Bearer |
Part of the Mission Critical Push-To-Talk (MCPTT) services. Used to trigger MCCP Map Group to Bearer Procedure. Available when MB2 interface is enabled.
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Establish a Group Call. SIP Supplementary Services prior versions do support A, B, C, D, E, F as a single entity like UE or Conference Server. In contrast, Landslide 18.8 extends meaning of B. It expects that B can be either a single UE or a group of UEs. B-as-a-group is supported by the Group_CallConnect script. Additional details in SIP Built-in Scripts - MSRP Close Group Chat An example of Script’s Callers distribution for Big Close Group is shown in the figure below:
Added support for "Range" of B-List-Subscribers for select Actions. If you edit an action to set the range to subs 1-100 and then edit the Caller Configuration to reduce # of subs to 66, when you click ok on the test case without adjusting the action you will get an error "Must select a B-Caller Subscriber Index 1 to 66 (Max Subs for B-List): See example below.
All – means all Bs specified @ “Configure Callers” (Figure above) are involved in the script. |
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Group_DataPathSetupMsrp |
Setup Data Path for MSRP Data Exchange. Note : As of release 19.4, we have changed the behavior of MSRP socket for Participants and conference server. Now MSRP socket for Participants are Active whereas MSRP socket for conference server is passive. This means when Participants are connecting to conference server, Conference server socket must be in running in passive mode. In short Conference server must run first. This action is responsible for Setup.
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Group_DataPathTerminateMsrp |
Terminate Data Path for MSRP Data Exchange. |
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Group_CallDisconnect |
Disconnect the Group Call. |
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Group_MediaStart |
Start Media within the Group. |
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Group_MediaStop |
Stop Media within the Group.
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Select to send Group data Management Message. This action will allow originator/participants to set/delete/move the Target to "Subject", "Bulletin" or "role" within the conference server.
XML editor functionality for MSRP messages GroupChat_SendLocationInfo (MSRP_SEND_LOCATION_INFO) and Group_DataManagement (MSRP_GROUP_DATA_MGMT) become available for IP Application Node with MSRP enabled.
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Available on the Mm Interface on PSAP Node Emulator Cnfg when E2 Interface, ESP is enabled. Use to Start the location query procedure on E2/ESP interface. Enter Source, Destination and Location of Device. Options : A, B,C,D,E,F,NET Select Device ID Type : Options : UUID, IPv4, IPv6, MAC, NAI, SIPURI, TELURI, FQDN, MSISDN, IMSI, IMEI, MIN, MDN Select Location Type : Options : Any, Civic, Geodetic, URI
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Select to Establish a connection (success, no auth) . This Action requires a previous call between source and destination. Added a configurable option to select the method for obtaining the Callback Destination Number (PSAP Node) : A callback Prefix String is optional and if provided (i.e. +1), PSAP will prefix it to a callback number (i.e. +1 ) to form a destination of call back. The 'Prefix String’ checkbox is unchecked by default; when checked, users can enter a prefix string (min=1,max=32,default=+1, any value except [:,@]). A Destination Number Preference is the following list of PSAP supported destination number types; Select one as the most preferable destination type for making a callback. When provided, PSAP will use a selected type for making a callback if that type is available; otherwise if a preference or a selected preference isn't available, PSAP will fall back to its own default selection. PSAP default destination preference is in this order: SIP P-Associated identity; then, SIP From username.
Per Spec Reference : NENA Standard for the the implementation of the Wireless Emergency Service Protocol E2 interface (NENA-05-001 December 2003) |
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Call_Back (success, auth req) |
Select to Establish a connection (success, auth req) . This Action requires a previous call between source and destination. Added a configurable option to select the method for obtaining the Callback Destination Number (PSAP Node) : A callback Prefix String is optional and if provided (i.e. +1), PSAP will prefix it to a callback number (i.e. +1 ) to form a destination of call back. The 'Prefix String’ checkbox is unchecked by default; when checked, users can enter a prefix string (min=1,max=32,default=+1, any value except [:,@]). A Destination Number Preference is the following list of PSAP supported destination number types; Select one as the most preferable destination type for making a callback. When provided, PSAP will use a selected type for making a callback if that type is available; otherwise if a preference or a selected preference isn't available, PSAP will fall back to its own default selection. PSAP default destination preference is in this order: SIP P-Associated identity; then, SIP From username.
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Call_Back (fail, no auth) |
Select to Establish a connection (fail, no auth) . This Action requires a previous call between source and destination. Added a configurable option to select the method for obtaining the Callback Destination Number (PSAP Node) : A callback Prefix String is optional and if provided (i.e. +1), PSAP will prefix it to a callback number (i.e. +1 ) to form a destination of call back. The 'Prefix String’ checkbox is unchecked by default; when checked, users can enter a prefix string (min=1,max=32,default=+1, any value except [:,@]). A Destination Number Preference is the following list of PSAP supported destination number types; Select one as the most preferable destination type for making a callback. When provided, PSAP will use a selected type for making a callback if that type is available; otherwise if a preference or a selected preference isn't available, PSAP will fall back to its own default selection. PSAP default destination preference is in this order: SIP P-Associated identity; then, SIP From username.
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Select to establish a connection. Use to Re-join call flow for the group call scenario. Important : Call_ReConnect action allows a user which has left the conference to join again by sending an INVITE to conference server using Group Session ID. This Group session-Id was received in INVITE request contact header earlier when the conference server had dialed out to a user. Design setup for ReConnect:
Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked.  
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Select to send MSRP Geo-Location.
XML editor functionality for MSRP messages GroupChat_SendLocationInfo (MSRP_SEND_LOCATION_INFO) and Group_DataManagement (MSRP_GROUP_DATA_MGMT) become available for IP Application Node with MSRP enabled.
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| Call_Redirect (success, no auth) |
Select for 302 Moved Temporarily response to the INVITE message. Available on IMS Nodal when Mw or ISC interfaces are enabled. Call ID Number is a positive number used to differentiate calls at a source or a destination. For instance, when a pair of source and destination has multiple calls, this option can be used for identifying a particular call. By default, this option is disabled. Range : 1 to 999999, 0 = unchecked. Flow:
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| Call_ForwardBusy_with_Reliable181 (success, no auth) |
Select to forward the call if calle is busy with Reliable 181 (SIP_RES_CALL_FORWARDED_181_Rel). In this action, Landslide will respond to 181 message with a PRACK. Also, Landslide will respond to 183 message with a PRACK. See the action details below.
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| Call_ForwardNoAnswer_with_Reliable181 (success, no auth) |
Select to foward the call if callee responsds but doesn't answer. In this action, Landslide will respond to 181 message with a PRACK. Also, Landslide will respond to 183 message with a PRACK. See the action details below.
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| SendText |
Used for sending a MSRP formatted message containing a simple plain text from one SIP client (source) to another SIP client (destination).
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| SendCpimMessage |
Used for sending a MSRP+CPIM formatted message containing a plain text from one SIP client (source) to another SIP client (destination). CPIM - Common Protocol for Instant Messaging
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| SendCpimFile |
Used for sending a MSRP+CPIM formatted message containing a pre-configured file from one SIP client (source) to another SIP client (destination). CPIM - Common Protocol for Instant Messaging
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| SendCpimFileInfo |
Used for sending a MSRP+CPIM formatted message containing a pre-configured file-info from one SIP client (source) to another SIP client (destination). CPIM - Common Protocol for Instant Messaging
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| HttpUploadFile |
Used for uploading a content of a pre-configured file to a remote file server using HTTP/HTTPS protocol.
If DNS is required for a test, configure DNS as shown…… when DNS is properly configured, the Test Server will query DNS servers for a real address of a given domain-name and replace a preconfigured destination file server with a real address received from DNS query.
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| HttpDnloadFile |
Used for downloading a content of a file from a remote file server using HTTP/HTTPS protocol.
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Document Number |
Title |
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RFC 5359 |
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RFC 3515 |
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RFC 3903 |
Session Initiation Protocol (SIP) Extension for Event State Publication |
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RFC 3428 |
Session Initiation Protocol (SIP) Extension for Instant Messaging |
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RFC 3261 |
Session Initiation Protocol (SIP) Session Initiation Protocol |
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RFC 5850 |
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RFC 4032 |
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RFC 6442 |
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RFC 3398 |
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ETSI GSM 5.0.0 |
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ETSI TS 24.008 |
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Website |
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3GPP TS 24.083 |
Call Waiting (CW) and Call Hold (HOLD) Supplementary Services |
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3GPP TS 24.615 |
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ETSI TS 124.604 |
Communication Diversion (CDIV) using IP Multimedia (IM) Core Network (CN) Subsystem |
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ETSI TS 124.628 |
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3GPP TS 23.280 release 14 |
Common functional architecture to support mission critical services |
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3GPP TS 22.179 release 14 |
Mission Critical Push-to-Talk (MCPTT) Stage 1 |
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3GPP TS 23.179 release 14 |
Mission Critical Push-to-Talk (MCPTT) Stage 2 |
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3GPP TS 24.379 release 14 |
Mission Critical Push-to-Talk (MCPTT) Call Control; Protocol specification |
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3GPP TS 24.380 release 14 |
Mission Critical Push-to-Talk (MCPTT) Media plane Control; Protocol specification |
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3GPP TS 33.180 release 14 |
Security of the mission critical service |
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3GPP TS 24.334 release 14 |
Proximity-Services (ProSe) User Equipment (UE) to PreSe function protocol aspects |
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3GPP TS 24.299 release 14 |
Session Initiation Protocol (SIP) and Session Description Protocol (SDP) |
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3GPP TS 24.238 |
SIP Based User Configuration - https://www.etsi.org/deliver/etsi_ts/124200_124299/124238/15.00.00_60/ts_124238v150000p.pdf |