Emulate and Test SIP for Next-Gen Networks

Welcome to the latest issue of GL's newsletter, featuring our SIP protocol emulation and testing solutions. In today’s fast-changing telecom environment, verifying that SIP-based VoIP devices and networks operate correctly and reliably is crucial. Our comprehensive test platform emulates all major SIP entities and generates real-time SIP and RTP traffic for extensive testing.

SIP protocol emulation and testing solutions

Overview

GL's Message Automation & Protocol Simulation (MAPS™) is a flexible software program that can emulate a wide variety of telecommunications protocols. MAPS™ SIP is a specialized application within this platform that focuses on testing SIP-based communication systems. It emulates key SIP elements such as User Agent Client (UAC), User Agent Server (UAS), Registrar, and Redirect servers.

The tool can emulate complex SIP call flows, supports high-volume call generation, and can transmit real-time voice, video, fax, and messaging traffic. With full support for SIP over UDP, TCP, and TLS, MAPS™ SIP helps verify system performance, troubleshoot interoperability issues, and ensure compliance with industry standards across VoIP and IP multimedia deployments.

Scalable Bulk Call Generation and Load Testing

MAPS™ SIP enables high-volume call generation for stress and capacity testing of SIP networks. Users can emulate thousands of concurrent voice or video calls using software-based configurations with automated call control and real-time monitoring. The tool supports up to 2,500 simultaneous media calls at 250 calls per second (CPS) and up to 70,000 concurrent signaling-only calls at 750 CPS.

For larger-scale testing, MAPS™ SIP integrates with the MAPS™ RTP High Density hardware appliance to generate up to 160,000 concurrent SIP calls at 800 CPS using specialized network interface cards. This setup helps assess network reliability by tracking key metrics like call success, failure, drop rates, mean opinion score, and packet loss under load. Furthermore, the MAPS™ RTP High Density can come in portable or rack-mount form factors. This makes it effective for both field and stationary testing.

Flexible SIP Emulation for VoIP and Air Traffic Control Networks

MAPS™ SIP can emulate any interface within a VoIP network, with a single instance capable of acting as multiple SIP entities simultaneously. It generates a wide range of SIP messages, minimizing the need for additional test equipment. The solution also supports VoIP implementations in line with the EUROCAE ED-137C standard, enabling the emulation of Air Traffic Control Communication networks.

SIP Testing for Gateway and Analog Telephone Adapter (ATA)

MAPS™ SIP evaluates SIP-based Gateway and ATA devices by testing call connectivity, signaling behavior, media traffic, codec support, and voice quality. It handles various RTP traffic types such as voice, digits, tones, pass-through fax, and T.38 fax making it suitable for functional, performance, and load testing across a wide range of SIP network scenarios.

Remote Control and Integration

The application supports remote operation through a Command Line Interface built on a client-server model. This allows users to control all application functions and automate test execution using Python or Java APIs. It enables seamless integration with external test systems and supports remote monitoring and control from different platforms, making it ideal for automated and distributed testing environments.

Fax over IP Emulation and Analysis

The software offers automated fax call generation and analysis for both T.30 and T.38 sessions, helping users verify transmission quality, protocol handling, and overall fax performance across SIP-based systems.

Instant Messaging Support for NG9-1-1 over SIP

MAPS™ SIP integrates with the Message Session Relay Protocol (MSRP) to support instant messaging over SIP in NG9-1-1 networks. It enables testing of various NG9-1-1 call scenarios, including IM-only sessions as well as combined audio and IM calls, helping validate end-to-end messaging functionality in emergency communication systems.

Multimedia Call Emulation with Audio, Video, and Text Messaging

MAPS™ SIP emulates multimedia SIP calls that include audio, video, and instant messaging. It transmits pre-recorded audio and video over RTP and exchanges text messages using MSRP within the same session. Each call setup includes three distinct media lines—one each for audio, video, and text messaging—enabling complete end-to-end testing of multimedia call handling in SIP networks.

Ensuring Protocol Compliance Across SIP Interfaces

MAPS™ SIP supports emulation of various SIP-based interfaces, including standard SIP, SIP-I (with ISUP signaling), SIP IMS, and SIP MSRP. It can execute SIP protocol conformance tests across diverse implementations.

End-to-End IP Multimedia Subsystem (IMS) Testing for VoLTE, VoWiFi, and 5G Services

MAPS™ SIP IMS Test Suite offers a complete solution for testing and validating signaling and media in IMS networks. It supports SIP, RTP, and Diameter protocols, and can emulate core IMS nodes like P-CSCF, I-CSCF, and S-CSCF. Ideal for VoLTE, VoNR, and multimedia service testing, it ensures smooth session management, mobility, and interoperability across LTE and 5G networks.

SIP Protocol Conformance Testing with ETSI Support

MAPS™ SIP Conformance Suite includes over 400 test cases based on ETSI SIP standards, covering key areas such as registration, call control, proxy behavior, and redirect server functions. It allows thorough validation of SIP protocol implementations by configuring the platform as a UAC to test UAS devices.

Automated Interactive Voice Response (IVR) Testing

MAPS™ SIP performs IVR testing by sending DTMF tones or voice input to interact with prompts and automatically navigate through IVR systems.


Key Features

  • Generates and processes SIP valid and invalid messages
  • Insert proprietary SIP headers at run-time
  • Implement IP security measures for SIP calls
  • Generates PCAP traces for SIP and RTP sessions
  • Supports both RTP G.711 pass-through fax and T.38 emulation over IP
  • 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
  • Adheres to the conformance test specification for SIP (IETF RFC 3261, ETSI TS 102-027-2 v4.1.1 (2006-07))
  • 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
    • 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, with transport over UDP, TCP, and TLS for secure communication. Also includes IPsec support for secure network-level transmission
  • Enables 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.
  • Support in dialog and out of dialog transactions for SUBSCRIBE, NOTIFY, OPTIONS, REFER, and INFO SIP methods
  • Supports 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 mean opinion score, packet loss, and jitter
  • Supports Short Message Service (SMS) over IP/ IMS communication. SMS is encapsulated in a SIP message and carried over the IMS core network

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