dsTest supports the Session Initiation Protocol (SIP) and Realtime 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 3G/4G/5G packet core networks.
dsTest maintains comprehensive sets of operational measurements for the SIPServer, SIPServer RTP, SIP Client, SIP Client RTP, SIP Socket and SIP IP layer. OMs are collected at configurable intervals and stored in a SQLite database on the dsTest server. Real-time measurements may be retrieved through our dsClient CLI interface or graphed via our dsClient GUI interface.
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
Enhance your SIP testing with Developing Solutions advanced testing features. Configure commands to define the series of actions and rates that each subscriber takes. Advanced testing profiles can be created using SmartEvents to define a mix of SIP traffic and other dsTest supported interface applications based on probability assignments.
- SIP – RFC 3261
- SIP Extensions – Various
- RTP – RFC 3550
- SDP – RFC 4566
- TCP – RFC 793
- UDP – RFC 768
General Reference Guides
- SIP Response Codes
- Diameter Dictionary
- Diameter Result Codes
- RADIUS Dictionary
- S1 Dictionary
- GTPv1 Dictionary
- GTPv2 Dictionary
- GTP Cause Codes
- Specification Map