SIP Editor

In VoLTE/IMS Node testing, the SIP tab on the Gm > Control tab allows you to define the transport used to establish, modify, and terminate SIP (Session Initiation Protocol) control protocol and modify content of the outgoing SIP messages. You may define a SIP message content using default templates or configure a custom SIP message format and/or a custom SDP message format.

The edit options available depends on whether you select Enable Supplementary on the VoLTE tab.

In AMF Nodal test case, a Gm2 tab is used for the Emergency Profile when Enable E911 Call is enabled.

SIP SCRIPT Mode: If you select Enable Supplementary on the VoLTE tab or IMS Node Emulator Configuration or Command Mode/Sequencer - About VoLTE ODC Sequencer and VoLTE OnDemand Commands.	You may edit the SIP flow to define customized SIP flows, actions, and messages or use the default/Built-in flows.
SIP Flow Mode: If you do not select Enable Supplementary on the VoLTE tab or IMS Node Emulator Configuration.	You cannot modify a flow as SIP is defined by a combination of the following enablers/controls on VoLTE tab: Enable Registration Enable Authentication Call Info

Transport

Enable TLS

Max Message Buffer Size (bytes)

SIP Flow:

(Enable Supplementary not selected on VoLTE tab or IMS Node, and NOT Command Mode/Sequencer)

SIP Messages

SIP Call Flow (Left pane)
SIP Message Editor (Right pane)

SIP SCRIPT:

(Enable Supplementary selected on VoLTE tab or IMS Node, or Command Mode/Sequencer)

SIP Supplementary Script - Create SIP tests using Scripts and Actions

SIP Message Editor (Right pane)

SIP Call Flow Editor (Left pane)
SIP Message Editor (Right pane)
Actions (Right Pane) - select from a list of actions
Built-ins (Right pane) - Select from a list of pre-built scripts

Command Mode/Sequencer - About VoLTE ODC Sequencer and VoLTE OnDemand Commands.

Transport

Select the type transport protocol used to establish, modify, and terminate SIP IMS sessions.

Options : UDP, TCP and UDP/TCP

When UDP/TCP is selected, UDP transport is used for the initial registration and all other messages via TCP transport.

NOTE: Either SIP Client or SIP Server can receive SIP Messages on a Transport Protocol independent of the transport protocol that was configured to send the SIP messages.

Tcl Parameter: SipTransport

Tcl Parameter: Gm2SipTransport

Enable TLS

Select to Enable TLS - Only supported when transport type is TCP. A new Tab TLS will be enabled where Cipher Suites, Private Key, and certificate can be used during TLS handshake.

Available on MME Nodal, SGW Nodal, IMS Node and IP application test cases.

Max Message Buffer Size (bytes)

Select to increase the message buffer size to avoid double fragmentation.

Available for all interfaces except Nc.

Range: 1024 to 10240

Default: 4096

Tcl Parameter: SipMaxMsgBufferSize

Tcl Parameter: Gm2SipMaxMsgBufferSize

SIP Messages

The SIP tab | SIP MessagesSIP Messages pane displays two columns/panes. The left pane displays SIP message flows line diagram and the right column/pane displays the relevant message header and message body content. When you click on a message (of a flow) in the left pane, the message is highlighted and on the right column/pane, displays the SIP message headers and SDP (SIP message body, where applicable) for the highlighted message.

NOTE:

The VoLTE application automatically generates and sends SIP messages. SIP messages are based on a pre-defined configuration. This configuration includes phonebook, P-header enablers and timing. Hence, applying user modification of SIP message contents is optional and it should be used only when you determine that auto-generated content does not fit a particular test case.

A default configuration is displayed as a template and allows you to modify SIP headers and body (SDP), where applicable. (A template is displayed for every SIP message available to be modified). The combination of the configurable SIP message format and the configurable SDP message format allows you to customize SIP messages to your SIP implementation (e.g., interoperability testing).

NOTES:

As of Release 15.0, a new design, uses a single Super-Flow with a fixed set of all possible messages.
The GUI will filter the display to only show the messages that apply to each specific configuration. The Super-Flow, Flow ID is 999.
All previously saved tests containing Flows with IDs 1-34 will be automatically upgraded to the new Super-Flow. When a Template is opened, you are able to select from all the Flow IDs and be automatically upgraded into the Super-Flow.
The Super-Flow has a display more aligned with how SIP Scripts display, but instead of "Actions" there are "Categories" such as "Registration", "Call", "Provisional Response", etc. As you enabled/disable features, the categories will be shown or hidden, and some messages inside the categories are also shown or hidden.
Unlike the old flows, all messages (shown and hidden) are always saved with the flow, and in Tcl API, the user has access to all the messages at all times. In the Tcl API, there is an "enabled" flag set upon validation of the test to indicate which messages actually matter. Only the messages that shown (GUI) / enabled (Tcl) are actually used by the TS.
Tcl API Scripts should be replaced with new Super-Flow as soon as possible.
Save-As-Tcl from the GUI will show how the new Super-Flow looks and/or ls::perform Validate on Tcl script to upgrade old Flow to Super-Flow.

Open a SIP Flow Template

Click to open the SIP Flow Template window and select a SIP Flow from the available libraries.

NOTE: The Save Template Dialog automatically uses the last opened/saved template name and you may edit it as required.

Save as Data Profile Template

Click to save the elected SIP Flow as a template.

NOTE: Copy of Data Profile and SIP Flow Templates prevent copy/move of system owned dependent items, e.g. {Basic}, {Scenario…} library items.

dock/undock

Pop-up SIP Flow

Click the dock / undock button, to open the SIP Flow in a larger re-sizable window.

SIP Flow Line Diagram

Click on any SIP Flow MessageSIP Flow Message (left column) to enable SIP Message editor (right column)

NOTE: If a SIP message is not available for editing, the right column remains blank. (Headers and SDP tab remain blank).

SIP Call Flow (Left Pane)

The SIP Call Flow (Left Pane) shows a flow of messages expected for a current test configuration. When you select a message in the flow control area, default SIP templates are shown in the editing windows (Right Pane).

The line diagram illustrates standard representation of the selected message. Each message (represented by a horizontal line) includes a text string (Field) or a combination of a text string and predefined parameter type (Filler; shown in blue on the right pane).

Paste Buffers

Use Paste Buffers to Edit SIP flow paste buffers used in super-flow message headers and SDP bodies. The Edit Paste Buffers window pops up to allow for capture of up to 10 paste buffers.

A new History dialog has been added to the Paste Buffer:

The left column indicated the Action-Message which includes : Action Id, Message # within the action and 'P' if part of Provisional Action.

Use Default

Selected by default. When selected, default content of SIP headers and SIP body are used in the test case.

When not selected, the template will be used to replace the default message content.

User Fields

Available if Use Default is not enabled.

Selecting User Fields provides you with the ability to insert a user defined custom headers, without any filler, for SIP / SDP Messages.

In the Tcl API, this attribute is UserFieldsEnabled.

##Get the default flow
set sip_flow_ [ls::perform GetDefaultSipFlow Gm 999 SipFlow $p_ ]
ls::config $sip_flow_ -UseDefault false
ls::config $sip_flow_ -UserFieldsEnabled true
ls::config $sip_flow_ -PasteBuffer0 "user#(N0)"
set msg1_ [ls::get $sip_flow_ -children-Message(0)]

SIP Flow Line Diagram

Click on any SIP Flow Message (left column) to enable SIP Message editor (right column)

NOTE: If a SIP message is not available for editing, the right column remains blank. (Headers and SDP tab remain blank).

SIP Messages Editor (Right Pane)

Use Default

When selected, for all messages in the test case, Use Default is disabled for individual message (on the Right Pane).

When not selected, individual message editor will be enabled for configuration.

Reset, Field+ (list), and Filler+ (list)

Available when Use Default is not selected. The Reset, Field+ , and Filler+Reset, Field+ , and Filler+ buttons help you with the editing process. You may replace the original Fields and Fillers in the message Header/Body (SDP) by any text.

Reset

Click to reset the modified content back to the default configuration

Field+

Click to display a list of text string. Select a text string to insert within your SIP message. You may also right-click at the insertion point, select the Insert SIP Field option from the context menu.

Filler+

Click to display a list of parameters/variables. Select a parameter to insert within your SIP message, which is shown in blue. You may also right-click at the insertion point and select the Insert Auto-Fill option from the context menu.

For example {User-text} is a SIP Filler that allows custom user text to be inserted into the SIP Message header.

See SIP Fillers for details.
See Body SDP Fillers for details.

Body Part +

Added the capability for XML editing (add or modify) for SIP Body (SDP).

Click to insert a body part template at current location.

Example edit :

Select Message on left hand side: REGISTER in this example.
Select Body Part to Insert @current Offset: 0 and mcptt-info+sml:3GPP 24.379 - Client Registration in this example
Clink OK to Insert message in Offset location.
Message is inserted.

NOTE: When pasting text in the message body, pay close attention to the CRLF Tokens

If you don’t want the CR in your selection, start your selection BEFORE the CR
If you want to include both CRLF, you must select from the line below i.e. after the real LF, not just the TOKEN
If you pick in the middle or the end of the CRLF tokens, then your paste will include the CR (without the LF)
If you paste in plain text from a different place, based on what you have enabled for line endings in the Message, the paste will automatically replace your clipboard text line-separators with enabled line endings (CR, LF, or CRLF) when it is pasted into the Message

CRLF/CR/LF

Click to insert carriage return (CR) or line feed (LF) characters or both (CRLF) at the selected byte/offset or right-click at the insertion point and select the New Line Mode option from the context menu.

Right-Click context menu

You may also right-click at the insertion point and select the required option from the context menu: Insert SIP Field, Insert Auto-Fill, or New Line Mode (CRLF, CR, LF).

You may also Open a QuickList and Add the Filler/Field to the QuickList for later use. The QUICKList function (RIGHT-CLICK menu) let’s you NAME your QuickList Messages. The QuickList saves the entire Message HEADER + BODY.

Editing SIP Message

Clear Use Default selected (both in the Left and the Right panes).
Select the required flow (represented by a line in the left pane), a pre configured template displays (where applicable) on the right pane.
Click the Header/Body (SDP) tab to edit, insert, modify, delete information by using the Field + (list) or the Filler + (list) as required.

Example of what happens when editing

Text string (field) that is not followed by a parameter (filler):

a=sendonly{CR}{LF}

Text string a=sendonly will always be included in the final message.

• Text string followed by a variable/parameter/filler:

Contact: <{Contact1-uri}>{CR}{LF}
If {Contact1-uri} exists in the original message then the final message will include string.

Contact: <sip:[email protected];transport=udp>

If {Contact1-uri} does not exist in the original message then the whole string will not be in the final message.

Contact: <sip:[email protected];transport=udp>

Conditional Parameters (used when parameters used in your template do not exist in the original message)

Since a configurable SIP message is generated from an existing message, some parameters used in your template may not exist in the original message. You may use “…-if” parameter to take care of such cases.

-if	Parameters with -if at the end are applied to the text string that starts from the previous parameter (Filler). Parameter with –if be inserted to indicate that this text string should be skipped if it does not exist in the message.
Examples:	a=sendonly{M2-a-sendonly-if}{CR}{LF} Text string a=sendonly will be included in the final message only if it exists in the original message.
	a=sendonly{CR}{LF} Attribute a=sendonly will be always included into the final message regardless of whether it exists in the original message.
	“Contact: <{Contact1-uri}>{Contact1-uri-if};isfocus{Contact1-isfocus-if}{CR}{LF}” {Contact1-uri-if} is used to include the right angle bracket > if the text string isfocus does not exist in the original message.

NOTE: For example, in the following string: Contact: <{Contact1-uri}>;isfocus{Contact1-isfocus-if}{CR}{LF}, if isfocus does not exist in the original message, the whole string >;isfocus will be missed/skipped and not included in the message.

Example SIP Header and Template

Example SDP

SIP Supplementary Script

The SIP Supplementary option, when enabled (Enable Supplementary) , 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 creating different SIP message flows tailored to their particular test environments.

A script (a sample is shown below) consists of one of more actions and a set of callers who will be responsible for performing actions of a script. Users have full controls of which actions or callers will be included in their scripts.

An action performs a particular small common task in SIP tests such as waiting for time, starting or stopping media, connecting or disconnecting users, etc. Some actions have predefined sequence of messages and to or from whom messages are supposed to be sent or received. Actions are fixed and predefined by Landslide software; they cannot be created by users.

Callers of a script refer to local subscribers and/or external subscribers will perform the script. For instance, a local subscriber #1 will play as Caller-A of a script and a local subscriber #2 will play as Caller-B of the same script. A common caller setting is that all odd subscribers (1, 3, 5, and so on) of a test will play a Caller-A of a script and all even subscribers (2, 4, 6, and so on) will play a Caller-B of the same script; however, any local or external subscribers can be chosen for callers of a script using the Landslide common auto-increment patterns (see Auto-Increment Help for detail).

See SIP Call Group Pre viewer for additional information.

Message

Paste Buffers

Use Paste Buffers to Edit SIP script buffers used in SIP Action message headers and SDP Body. Up to 10 paste-buffers.

In the example listed below:

aser will be placed in temp1: from {paste-buffer-1}
bser will be placed in temp2: from {paste-buffer-2}
cser will be placed in temp3: from {paste-buffer-3} in the INVITE Header message.

A new History dialog has been added to the Paste Buffer:

Copy-to-Buffers

This feature applies to Gm Supplementary Scripts. You can copy Fillers to the Paste Buffers, similar functionality to DMF copy/paste buffers.

The Copy-to-Buffers becomes available if your message is edit-able.

Select any Filler to be copied to any buffer. That same filler must be defined/present in the message.

Double click on Copy Filler for Insert Filler pop-up:

Add text to Paste Buffers:

Copy Text From Paste Buffer:

Validation rules for:

Each buffer can only be copied to once per Message, in either Header or Body part.
Within a single message, you can either copy to or paste from a buffer, but not both.
You can paste multiple times from same buffer, unlike rule #1 for copies.
You cannot copy a filler that is not being used by the message.

When you OK the Edit Copies-to-Buffers dialog only Rule #1 will be checked.

When you OK the test case, all rules will be checked.

Some examples of errors that fail the rules:

Note the difference in the labeling, if you don’t get the “Action”/”Message” detail in the error message, it is referring to the currently selected Message.

A History Dialog has been added to the Paste Buffer:

NOTEs:

One of the limitations of Copy-Paste buffer on Sending side when sending a custom SIP-Configurable message, is that case the SIP-Filler that would be copied will be one that is produced by “Use Default” message not the one produced by SIP-Configurable message. eg: If the user wants to copy From-URI into Paste-buffer-1 from the message INVITE being sent out. From the phonebook, lets say default Sip produced is From-URI : [email protected]. However user changed the From-URI to [email protected]. In that case Paste-buffer-1 will contain value [email protected] and not [email protected]
If you put a COPY inside of a message:
- For messages that are being sent, you are just copying the data generated by your test case.
- For messages that are being received, you are copying the value received from the other side.
Or if you use the same script template in multiple locations, only the messages that are received in perspective to your test case, will be actually doing the copy from live data. Any messages being sent are fully produced by your local Test Case.
Or if you are emulating all of the Subscribers/Endpoints in one test case, the COPY feature is not very useful

Use Default

Selected by default. When selected, default content of SIP headers and SIP body are used in the test case.

When not selected, the template will be used to replace the default message content.

User Fields

Available if Use Default is not enabled.

Selecting User Fields provides you with the ability to insert a user defined custom headers, without any filler, for SIP / SDP Messages.

In the Tcl API, this attribute is UserFieldsEnabled.

Actions

See Complete list of Actions - Sip Actions

Built-ins

See Complete list of Built-in Scripts - Sip Built-in Scripts

Send Delay (ms)

Select to support a delay at the message level for Supplementary Scripts. The SIP signaling messages will have an option to specify a send delay.

Range : 0 to 65535 milliseconds

Default : 0 which means no send delay

The Send Delay option will be driven by the "delay" attribute in the SIP.xml files:

If the delay attribute is missing, or if the value is -1, the Message will NOT support Send Delay.

Example:

<Action prototype-id="1" class-id="1" odc-only="true">
<Notes>Call_Connect (success, no auth)</Notes>
<Notes>Establish a connection</Notes>
<Role label="A" party-id="1"/>
<Role label="B" party-id="2"/>
<Role label="NET" party-id="254"/>
<Property name="CALL_ID_NUM" value="0"/>
<Message from="A" to="B" msg-type="SIP_REQ_INVITE" msg-prototype-id="1" default-id="1"/>
<Message from="B" to="A" msg-type="SIP_RES_INVITE_2xx" msg-prototype-id="2" default-id="6" time-offset="0"/>
<Message from="A" to="B" msg-type="SIP_REQ_INVITE_2xx_ACK" msg-prototype-id="3" default-id="14"/>
</Action>

Limitations:

The delay timer is applied to Landslide sending messages. We do not delay processing of received messages.
Currently, the Send Delay configuration applies to only SIP Signaling messages.
There will be no Send Delay configuration for media actions, e.g.: Media_Start/Media_Stop_Media/Media_wait/Media_SendWait_/GroupChat_Send.
Existing tests will not be affected by this feature. Send Delay applies to new Actions. For any existing saved tests, the user will have to delete the existing Action and re-add the Action back to get the latest definition including the Send Delays for the messages.
The delay cannot be applied to CALL_DISCONNECT/FAIL/GROUP_CHAT/MCPTT action. In case of Failure, some of the message are triggered by RADVISION automatically without consultation with Supplementary flow.
There might be cases where delay is not applied due to RADVISION or code path may be different because of call backs

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SIP Call Group Pre viewer

A Call Group is defined as a group of subscribers and/or endpoints that take part in a SIP Script instance or call together.

When you configure SIP Scripts you assign Subscribers and Endpoints to Callers in your Script. When the test runs, the associated Subscribers and Endpoints take part in an execution of the instance of the SIP Script.

In this example, A and B are callers. When an instance of this script runs two specific Subscribers are taking part in a call, one as A and one as B:

The # of Subscribers in a test case = # of UEs * # of Subscribers Per UE. And the # of Endpoints == # of Subscribers if Endpoint is enabled (IMS Node).

The redesigned Configure Callers dialog provides for an easier test setup, UE-Subscriber assignment understanding, and added a new field “# of B Subs” for the B-Caller in the Caller Configuration:

The separate indicators for “# of Callers” has been removed, as this really represents how many instances of your Script or Call (we call these Caller Groups) will be executed. This is a single value and determined by the limiting factor in your caller configuration. If you have 4 Callers in your script, A,B,C,D, and you have 100 Subscribers in your test, at best you can run 25 Caller Groups, 4 Subscribes in each Script. Within a test case, you can have some callers as Endpoints and some as Subscribers, so these are separate sets. If the previous example was 3 Subscriber Callers and 1 Endpoint Caller, at best there can be 33 Caller Groups, but you will only use 33 Endpoints out of a possible 100, because the 3 Callers must be shared across 100 Subs.

Open Conference Group : As of Release 19.0, we added the ability to add up to 100 users in a conference call (Number of Callers in SIP Supplementary Service Tests ).

Open Conference Group. : Open Group is based on SIP REFER message (A.2.1 @ 3GPP TS 24.605).

Open Group Scenarios rely on A/B/C/D/E Callers as UEs and Caller F as CONF. It has been changed in the following way:

Caller A – UE
Caller B – UE or a group of UEs: B1-BN (1 ≤ N ≤ 100)
Caller C – CONF (if Conference Server is a part of test scenario)

If, in fact, N = 8 then it will be a Caller Group of 10: 9 UEs and 1 CONF (if needed).

It does not mean that existing CALLERs D/E/F cannot be used. But it seems a group of A/B/C is powerful enough to provide any script scenarios. From the other end, a script can be limited B-Caller group only: i.e. instead of having A, B, C, D, E, F in group we can use B1, B2, …, Bn.

Two actions, bellow, illustrate how it can be used:

Action Call Connect: A calls B7

Action Call Refer: A refers B9 to C (CONF)

B-Callers as a group of up to 100 Subscriber or Endpoints. This concept is applied to ALL actions: Call Connect, Call Disconnect, Call Refer, Media Start, Media Stop, and so on…

Configure Callers: B-Caller as a group of 9

and allow to specify a Sub Caller in B-Caller in any action:

“Configuring Callers” specifies a list of X-Callers participating in a Supplementary Service script. B-Caller can be either a single Subscriber or Endpoint or a group of Subscribers or Endpoints, Sub-Callers. Number of Sub-Callers is a number within the range of 1-100.

An example of B-Caller-as-Group is shown here:

In addition in Release 18.8 with a special license, we added the ability to have the B-Caller be a list of UE-Subscribers or Endpoints, so that within a Call (Caller Group), you can have Caller B represent up to 100 Subscribers/Endpoints. This feature would only apply to certain special Actions, so while you might set the value to 100 in the Caller Configuration, most Actions would still have to pick just one Subscriber out of the list.

An Auto Fill Wizard is available to help setup your scripts to use Subscribers in Round-robin or Sequential order. With a special license where B-Caller be a list of UE-Subscribers or Endpoints, so that within a Call (Caller Group), you can have Caller B represent up to 100 Subscribers/Endpoints. To be used with Special Actions/Scripts : SIP Built-in Scripts (MSRP Close Group Chat), SIP Built-in Actions (Group_CallConnect, Group_DataPathSetupMsrp, Group_DataPathTerminateMsrp, Group_CallDisconnect, Group_MediaStart, Group_MediaStop),

See example below: # of B Subs = 100

The Auto-filler will automatically find the maximum number of Caller Groups based on the number of Subscribers available in the test case.

If you use the wrong auto-incremented configuration, you can end up with overlaps of a single Subscriber or Endpoint being used in the same instance or even other instances.

You will get an indication of what is the limiting factor for any configuration.

The Callers Previewer has been combined with the Callers Configuration to enable you to configure your SIP Script Call Groups and to see possible duplication:

You do however, need to click the "Preview-Refresh" button to refresh the preview after you change your configuration.

Figure Below shows Overlap Error:

Correct Overlap error by Using Auto-filler option : ABCABCABC

Or Correct Overlap error by Using Auto-filler option : AAAABBBBBBCCCCC :

Switch C from Subscriber to Endpoint and use the Auto-Fill Wizard to configure callers:

The # of Call Groups will be determined as follows:

NumCallGroups = Number of Subscribers;

For each Caller

If # of Callers < NumCallGroups

NumCallGroups = # of Callers

int callers = 0

for each Caller Index in auto-incremented Caller value

callers++

if Caller index > number of Subscribers

NumCallGroups = callers

break;

Examples:

1. # of Subscribers in test = 1000

Caller A #(N1) 500 Subscribers
Caller B #(N501) 500 Subscribers
Number of Call Groups = 500 Limited by # of Callers = 500 for Caller A/B

2. # of Subscribers in test = 1000

Caller A #(N1) 500 Subscribers
Caller B #(N501) 400 Subscribers
Number of Call Groups = 500 Limited by # of Callers = 400 for Caller B

3. # of Subscribers in test = 1000

Caller A #(N1) 500 Subscribers
Caller B #(N801) 500 Subscribers
Number of Call Groups = 200 Limited by Caller Index > 1000 in Caller B (801-1000)

4. # of Subscribers in test = 100

Caller A #(N1) 500 Subscribers
Caller B #(N801) 500 Subscribers
Number of Call Groups = 0 Limited by Caller Index > 100 in Caller B (801-1000, can’t be done)

5. # of Subscribers in test = 100

Caller A #(N1 0/2) 500 Subscribers
Caller B #(N2 0/2) 500 Subscribers
Number of Call Groups = 100 Limited by Number of Subs in test > 100

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