Phone: (469) 634-4200

dsTest, Developing Solutions, SIP, SIPServer, SIP Testing, IMS Testing, VoLTE Testing

dsTest supports the Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP), implementing RFC 3261 (base SIP) with various RFCs that add SIP extensions. SIP is one of the 3GPP signaling protocols, utilizes the Session Description Protocol (SDP), and is a part of the IMS architecture. SIP is very similar to HTTP in that each transaction consists of a client request that invokes a particular method or function on the server and at least one response, and reuses most of the header fields, encoding rules and status codes of HTTP.

With dsTest’s set of EPC interface applications, such as  Rx, Cx/Dx, Sh/Dh, Ro/Gy, and Rf/Gz, and our SIP interface signalling simulation between IMS nodes, dsTest provides full featured Voice over LTE (VoLTE) testing  for end-to-end testing capabilities in 4G packet core networks.

Read more about our VoLTE test solutions.

Implementation

dsTest allows the user to configure and send the following SIP methods and response messages:

  • INVITE – indicates that the user or service is invited to participate in a session
  • ACK – final confirmation and concludes the transaction initiated by the INVITE command
  • REGISTER – location service for user agents, indicating their address information to the SIPServer.
  • BYE – release a call.
  • CANCEL – cancel any in-progress request
  • PRACK – Provisional acknowledgement.
  • PRACK Complete – (Response Code 200 for PRACK)
  • UPDATE – Modifies the state of a session
  • Answer (Response Code 200)
  • Trying (Response Code 100)
  • Ringing (Response Code 180)
  • Progress (Response Code 183)

A complete set of events that can be generated with the dsTest SIP Interface application can be found here.

As part of the dsTest SIP interface application, you can configure RTP element and attributes, including:

  • Configure SIPServer Media attributes – type, port, media format attributes
  • Configure SIP User Plane Data
  • Specify RTP Client configuration
  • Enable/Disable RTP data
  • Configure RTP data interval
  • Configure RTP media attributes – audio/video, codec type, sampling rate

Advanced Testing Features

SmartEvents — Alter application behavior or coordinate multiple interface applications with SmartEvents. Our programmable, subscriber-level state machine gives you the ability to define handlers for specified application events that can then cause an event in another application to be executed. You can also configure handlers to modify subscriber information during run-time, introduce timers, or randomize subscriber behavior based on configurable probabilities to name a few of the many options in one of dsTest’s most powerful features.

Traffic Profile — Draw the shape of your test actions across time with Traffic Profile. You can define the rate for any action as a static rate or reference a Traffic Profile configuration, which also means that multiple Traffic Profiles can be running currently. Use Traffic Profile in conjunction with the randomizing features in SmartEvents to design a test that more truly simulates real-world network activity.

Operational Measurements

dsTest provides rich sets of measurements for the SIP application:

  • Transaction and transport layer attempts, successes, and failures
  • Transaction duration, transactions-per-second, and round-trip delay
  • Message and byte counters
  • Errors encountered and error indications received in messages

See our Online Help for SIP measurements, RTP measurements, and socket measurements.

You can read more about the reporting features offered with dsTest and dsClient here.

Reference Standards

RTP StackdsTest, Developing Solutions, SIP, SIP Testing, IMS, SIPServer