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SIP User Documentation Chapter 5

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Hemanth Kumar Vulavala edited this page Aug 17, 2020 · 2 revisions

Adding and configuring SIP nodes

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Add the SIP node element to your Test Plan as you add other Config Elements in JMeter. Define the SIP Node name, SIP Node context parameters, and SIP Stack endpoint parameters.

Adding SIP nodes to the Test Plan

Procedure
  1. Start Apache JMeter in GUI mode.
  2. Right-click Test Plan and select Add → Config Element → SIP node.
    You can emulate several network elements by adding several SIP nodes to the Test Plan.

Configuring SIP nodes

Procedure
  1. Add a label and a name for the SIP node. *_ Name_
    Add the label that appears in the Test Plan tree structure. This field is not mandatory but it is recommended to fill it.
  • Comment
    Add a short description of the SIP nodes. This field is not mandatory.
  • Node name
    Add the SIP node name that must be referenced in other elements of the Test Plan. This string must be unique among all configuration elements in the Test Plan.
  1. Configure the global SIP nodes context parameters in the Stack parameters box.
  • traffic type
    • Select the FUNCTIONAL option to use the SIP nodes in functional mode.
    • Select the PERFORMANCE option to use the SIP nodes in performance mode.

NOTE: For more details on traffic type, see SIP node traffic types page.

  • Automatic answer delay (only used in Performance mode)
    Delay in milliseconds when an incoming request is automatically answered. This property is used only in performance mode.
  • SIP message reception timeout
    Specify the SIP message reception timeout in milliseconds. The default value is 5000 milliseconds. If a sampler that is configured to check the reception of a specific request or response cannot find the message immediately in the reception queue, it waits until this timeout period before reporting that the message was not received.

NOTE: The default value of 5000 milliseconds is used by the node even if the SIP message reception timeout field is empty.

  • Session max life time
    Add the maximum lifetime for SIP messages in milliseconds. The default value is 30000 milliseconds. You can use a variable here, for example, <$(session-life-time)>.

IMPORTANT: This field is very important to avoid local dialog leakage in case of a remote application issue.

  • outbound proxy of the SIP Stack : ipaddress:port/transport
    If you want to specify an outbound proxy, select Edit from the drop-down menu, and add the address of the outbound proxy between the simulated element and the SIP network. The format is <ipaddress>:<port>/<transport> where <transport> refers to the transport layer used for the connection.

  • Ignore retransmission for non reliable answers

    If you want to specify an Ignore retransmission for non-reliable answers, select Edit from the drop-down menu.

    • Select the Yes option to use the forked non-reliable response is a retransmission and retransmission are ignored
    • Select the No option to use the forked non-reliable response is a retransmission and retransmission are not ignored

  • Ignore retransmission for reliable answers

    If you want to specify an Ignore retransmission for reliable answers, select Edit from the drop-down menu

    • Select the Yes option to use the forked reliable response is a retransmission and retransmission are ignored
    • Select the No option to use the forked reliable response is a retransmission and retransmission are not ignored

  • Automatic queues cleanup

    If you want to specify an Automatic queues cleanup, select Edit from the drop-down menu

    • Select the Yes option to use the dialog list (sessions) are cleaned when SIP node state(before start of scenario’s) is ready
    • Select the No option to use the dialog list (sessions) are not cleaned when SIP node state(before start of scenario’s) is ready
  1. Configure the SIP Stack endpoint parameters.

  • local IP address
    Add a specific local IP address or use a variable.

  • localPort
    Add a specific local port or use a variable.

  • transport layer
    Select the UDP, TCP or SCTP communication protocol as the transport layer for the SIP connection.

  • Maximum number of SIP connections

    Max number of simultaneous TCP connections handled by stack

  • Is caching of SIP connections enabled

    This parameter is not used

  • Header name
    Select the header name to be used to identify a call in case of incoming calls.

    • To
    • From
    • Request-URI

Depending on the service you want to test, you can sort the incoming calls based on the To, From or Request-URI header value.

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© Copyright 2019-2020 Hewlett Packard Enterprise Development LP

Licensed under the Apache License, Version 2.0 (the "License"); you may not use this file except in compliance with the License.
You may obtain a copy of the License at http://www.apache.org/licenses/LICENSE-2.0.

Unless required by applicable law or agreed to in writing, software distributed under the License is distributed on an "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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