Gaithersburg, Maryland, USA - October 01, 2024, GL Communications Inc., a global leader in telecom test and measurement solutions, addressed the press regarding their SIP Protocol Emulation and Testing Solutions. These solutions offer a comprehensive framework for all aspects of SIP signaling and voice testing for a variety of network environments.

Vijay Kulkarni, CEO of GL Communications, states, “GL's Message Automation & Protocol Simulation (MAPS™) is a versatile software program that emulates various telecommunications protocols. MAPS™ SIP emulates User Agents, Proxy, Redirect, Registrar, and Registrant servers. This tool can place and answer calls, replicating the behavior of an actual VoIP phone. All call parameters are customizable, including calling number, called number, duration, audio payload and more.”

MAPS™ SIP seamlessly integrates with the MAPS™ RTP HD hardware appliance to generate tens of thousands of simultaneous calls through specialized network interface cards. This makes it ideal for load testing network infrastructure. It reports metrics such as successful calls, failed calls, dropped calls, mean opinion score and packet loss, testing the robustness and reliability of the network under heavy traffic conditions.

MAPS™ SIP can emulate any interface within a VoIP network, with a single instance capable of functioning as multiple SIP entities simultaneously. It generates various SIP messages, reducing the need for additional testing equipment, and supports VoIP implementations in compliance with ED-137C of the EUROCAE standard, enabling the emulation of Air Traffic Control Communications networks.

The SIP testing solution evaluates Gateway and ATA products, covering key areas such as call connectivity, call signaling, traffic generation, voice quality testing and codec functionality. It transmits and detects various RTP traffic types, including digits, voice files, pass-through fax, T.38 fax, single tones, and dual tones, and can generate up to 2,000 simultaneous RTP media calls at a rate of 250 calls per second.

The software can handle 70,000 concurrent calls at 750 calls per second for SIP signaling-only scenarios and features a Command Line Interface, allowing users to control all functionalities through Python and Java APIs. It offers automated fax call emulation and analysis for T.30 and T.38 fax sessions, enabling efficient testing and analysis of fax transmissions over IP networks.

Additionally, it integrates with the Message Session Relay Protocol (MSRP) to facilitate instant messaging over SIP sessions in NG9-1-1 networks, supporting various NG9-1-1 call types, including instant messaging (IM)-only, audio and IM, and video and IM calls across multiple user agents. MAPS™ SIP also performs Interactive Voice Response (IVR) testing, recognizing and responding to voice prompts using DTMF digits or voice for automated IVR traversal.

GL’s SIP testing tool excels in emulating diverse interfaces within a SIP network, including standard SIP, SIP-I (ISUP), SIP IMS, and SIP MSRP, while performing comprehensive SIP Protocol Conformance Testing for various SIP implementations. The SIP protocol is integral to ED-137, the Next Generation Air Traffic Communications Protocol.

The MAPS™ SIP IMS Test Suite provides a robust platform for testing, simulating, and validating signaling and media interactions within IMS networks. Supporting key protocols such as SIP, RTP, and Diameter, it ensures seamless communication across LTE and 5G. With the capability to emulate key IMS nodes like P-CSCF, I-CSCF, and S-CSCF, MAPS™ for IMS is ideal for testing end-to-end call flows, mobility scenarios, and session management.

GL’s MAPS™ SIP Conformance Suite features over 400 test cases aligned with the SIP specifications from the European Telecommunications Standards Institute (ETSI) TS 102 027-2 V3.1.1 (2004-11). The suite adheres to essential protocol standards for assessing the conformance of SIP servers and client entities. It includes built-in scripts designed for Proxy, Redirect Server, Registrar, UAC, and UAS conformance.

Key features

• Generates and processes SIP valid and invalid messages
• Insert proprietary SIP headers in run-time
• Implement IP security measures for SIP calls
• Generates PCAP traces for SIP and RTP sessions
• Efficient handling of RTP media on remote systems
• Logs SIP call messages along with their decoding for every call
• Automation for the auto call rejection feature, supporting sequential/random error codes
• Manages strict routing and loose routing for requests routed through proxies
• Adheres to the conformance test specification for SIP (IETF RFC 3261)
• Configure display names for Contact, From, and To Headers
• Configure INVITE Expiry, offering an alternative method for determining call progress timeout
• Supports IPv4 and IPv6 and transport over UDP and TCP, and TLS for secure transport
• Supports conference calls, unattended call transfer, attended call transfer, call hold, auto call rejection, early media, and silence packets generation
• Implement IP Spoofing for any network like Class C, Class B, etc.
• Supports in dialog and out of dialog transactions for SUBSCRIBE, NOTIFY, OPTIONS, REFER, and INFO SIP methods
• Supports industry-standard codec types - G.711 (mu-Law and A-Law), G.722, G.729, G.726, GSM, AMR, AMR -WB, EVRC, EVS, OPUS, SMV, iLBC, SPEEX, and more
• Supports Secure Real-time Transport Protocol
• Provides voice quality statistics such as MOS, packet loss, and jitter
• Supports both RTP G.711 pass-through fax and T.38 emulation over IP
• Supports Short Message Service (SMS) over IP/ IMS's communication. SMS is encapsulated in a SIP message and carried over the IMS core network

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