GL’s Message Automation & Protocol Simulation (MAPS™) designed for SIP testing can simulate User Agents (User Agent Client- UAC, User Agent Server-UAS) Redirect and Registrant servers. This test tool/traffic generator can be used to simulate any interface in a SIP network and perform protocol conformance testing (SIP protocol implementations).
The application is available as:
The application gives the users the unlimited ability to edit SIP messages and control scenarios (message sequences). “Message sequences” are generated through scripts. “Messages” are created using message templates.
MAPS™ can be used to simulate any interface of the VoIP network. Single MAPS™ instance can act as more than one SIP entity at a time and can generate any SIP message on wire in VoIP network and hence equipment needed to test are reduced. MAPS™ SIP supports UDP, TCP, TLS transport types. Secure Real-time Transport Protocol (or SRTP) traffic supported over TLS (Transport Layer Security) Transport and TLS Library (OpenSSL) uses a Certificate and Key.
With the purchase of RTP Core license (PKS102), MAPS™ SIP supports transmission and detection of various RTP traffic such as, digits, voice file, single tone, dual tones, IVR, FAX*, and Video*. With regular RTP traffic, the maximum Simultaneous Calls up to 2000, and Calls per Second up to 250 is achievable. Almost all industry standard voice codec supported.
The MAPS™ SIP Conformance Scripts (PKS121) is designed with 400+ test cases, as per SIP specification of ETSI TS 102-027-2 v4.1.1 (2006-07) standard. Test cases include general messaging and call flow scenarios for multimedia call session setup and control over IP networks. Logging and pass/fail results are also reported. Test cases verify conformance of actions such as registration, call control, registrants, proxies and redirect servers.
MAPS™ SIP also supports generation of high volume of calls with traffic for load testing network using MAPS™ RTP HD (PKS109) network appliance.
MAPS™ RTP HD (requires a special purpose network appliance and PKS109 RTP HD licenses) is a specialised 1U rack mounted network appliance designed to easily achieve up to 32,000 endpoints per appliance (8000 simultaneous calls with duplex traffic per port). The appliance is available with 4x 1 Gbps NIC ports. Using a stack of multiple servers, a larger test system with 100K-200K calls (all controlled from a single Master Controller) is achievable for enterprise to carrier grade testing.
GL also provides a high-density network monitoring appliance (supports SIP, RTP, UMTS, LTE, and IMS) for online capture and decode of the signalling in real time.
** Some of these traffic types requires additional licenses – get in touch for more information.
RTP traffic simulation is supported for almost all standard codecs. Also, supports various traffic events simulation during the course of a call, which is listed below:
Signalling Events
RTP Traffic Events – digits, tones, files
MAPS™ SIP provides the Bulk Video Call Simulation capability using pre-recorded video traces supporting codecs like H.264, H.263, and VP8. On a high-performance computing platform (core-i7), it is possible to generate more than 500 simultaneous video calls. It also provides statistics for the RTP traffic such as Listening MOS, Conversational MOS, PacketLoss, Discarded Packets, Out of Sequence Packets, Duplicate Packets, Delay and Jitter.
MAPS™ SIP supports FAX over IP (FoIP) simulation and monitoring. With Additional licensing, both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211) simulation are supported.
MAPS™ SIP supports Message Session Relay Protocol for instant messaging over SIP sessions (PKS112), simulating SIP/MSRP User Agents end-points in an NG9-1-1 network and send and receive communications over ESInets. The NG9-1-1 calls can be IM Only Calls, Audio and IM Calls, and Video and IM Calls between multiple UAs.
MAPS™ SIP supports Interactive Voice Response (IVR) testing that recognises and responds to voice prompts using DTMF digits or voice, allowing automated IVR traversal and testing.
MAPS™ SIP handles simulation of Audio and Video along with IM call, during which the pre-recorded audio/video is sent and received using RTP and text messages using MSRP during the same call. During this call type, there will be three media lines, one for Audio, one for Video and another for text messages using MSRP.
MAPS™ SIP also supports command line interface to remotely control MAPS™ from client environment. Multiple MAPS™ CLI servers from the client environment (such as TCL, Python, VBScript, Java, or .Net) via APIs for complete automation.
GL also provides a Packet Analyser for on-line capture and decode of the SIP signalling in real-time both during tests and as a stand-alone tracer for live systems.
If you’re interested in the products shown here, or have questions about something you can’t find on our website, please get in touch through the form below.
Our knowledgeable sales team will be happy to help. We can normally source exactly what you need, or work with you to come up with custom solutions.
Alternatively, just call +44 (0) 1933 461 666 and ask to speak to the networks team.
