Advanced SIP Protocol Testing Solutions

Welcome to the latest issue of GL's Newsletter providing insight into our 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.

MAPS™ SIP Protocol Emulator

Overview

GL's Message Automation & Protocol Simulation (MAPS™) is a versatile software program that can emulate a wide variety of telecommunications protocols. MAPS™ SIP can emulate User Agents (User Agent Client - UAC, User Agent Server - UAS), 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.

High-Volume Call Emulation with MAPS™ SIP and RTP HD Integration

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

Versatile SIP Emulation for VoIP and Air Traffic Control Networks

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

Comprehensive SIP Testing for Gateway and Analog Telephone Adapter (ATA) Products

The SIP testing solution can evaluate Gateway and ATA products, encompassing key areas such as call connectivity, call signaling, traffic generation, voice quality testing, and codec functionality, along with numerous other features. The tool transmits and detects various RTP traffic types, including digits, voice files, pass-through fax, T.38 fax, single tones, and dual tones. MAPS™ SIP can generate up to 2,000 simultaneous RTP media calls at a rate of 250 calls per second (CPS). The application can handle 70,000 concurrent calls at 750 CPS for SIP signaling-only scenarios.

Enhancing Automation with MAPS™ SIP's Command Line Interface

The software features a Command Line Interface (CLI) based on a client-server model, allowing users to control all functionalities through Python and Java APIs.

Automated Fax Emulation and Analysis for T.30 and T.38

The software 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.

Advanced Messaging Capabilities in NG9-1-1 Networks

The software integrates with the Message Session Relay Protocol (MSRP) to facilitate instant messaging over SIP sessions in NG9-1-1 networks. It supports various NG9-1-1 call types, including instant messaging (IM)-only, audio and IM, and video and IM calls across multiple user agents.

Efficient IVR Testing and Automation

Additionally, MAPS™ SIP performs Interactive Voice Response (IVR) testing, recognizing and responding to voice prompts using DTMF digits or voice for automated IVR traversal.

Multimedia Call Emulation with MAPS™ SIP

MAPS™ SIP emulates multimedia calls, incorporating audio, video, and instant messaging. It transmits pre-recorded audio and video using RTP while exchanging text messages via MSRP within the same call. This setup involves three media lines: one for audio, one for video, and another for text messages using MSRP.

Ensuring Protocol Compliance Across SIP Interfaces

Our SIP testing tool excels at emulating diverse interfaces within a SIP network (standard SIP, SIP-I (ISUP), SIP IMS, and SIP MSRP) and execute SIP Protocol Conformance Testing for varied SIP protocol implementations. The SIP protocol is also integral to ED-137, the Next Generation Air Traffic Communications Protocol.

IP Multimedia Subsystem (IMS) Testing for LTE and 5G Networks

MAPS™ SIP IMS Test Suite provides a robust platform for testing, simulating and validating signaling and media interactions in IMS networks. Supporting protocols like SIP, RTP, and Diameter, this solution ensures seamless communication over LTE and 5G. With the capability to emulate key IMS nodes such as P-CSCF, I-CSCF, and S-CSCF, MAPS™ for IMS is perfect for testing end-to-end call flows, mobility scenarios and session management. Whether for VoLTE, VoWiFi, or multimedia services, MAPS™ for IMS helps operators and manufacturers ensure quality, reliability, and interoperability across their IMS networks.

SIP Conformance Testing with MAPS™ SIP Suite

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). This suite adheres to the essential protocol standards for assessing the conformance of SIP servers and client entities. It includes built-in scripts tailored for Proxy, Redirect Server, Registrar, UAC, and UAS conformance, facilitating comprehensive testing in accordance with the ETSI standard.

The test cases cover general messaging and call flow for multimedia call session setup and control across IP networks. The testing process includes logging and reporting of pass/fail results. These cases meticulously verify conformance in key areas such as registration, call control, proxies, and redirect servers.


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
  • Automates T.30 and T.38 fax 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)
  • UA behavior involves:
    • User Agent Client initiating requests
    • User Agent Server responding to requests
  • Entities and their roles include:
    • Redirect Server: User Agent Server redirecting requests
    • Proxy: Making requests on behalf of other clients
    • Registrar: Accepts REGISTER requests
    • Registrant: Sends the REGISTER message
  • 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 almost all 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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