GL Enhances MAPS™ ED-137 Telephone Emulator for VoIP Air Traffic Management

Welcome to the latest issue of GL's Newsletter providing insight into our voice communication solution MAPS™ ED-137 Telephone for VoIP Air Traffic Management (ATM). In this newsletter, we highlight the important enhancements for MAPS™ ED-137 Telephone software version 22.6.14.

MAPS™ ED-137 Telephone Emulator


GL’s MAPS™ ED-137 Telephone Emulator can emulate Ground-to-Ground calls as per EUROCAE ED-137B Volume 2 and ED-137C Volume 2 Telephone standards. The software provides complete control over call scenarios and the ability to customize network parameters for signaling and VoIP traffic. The software can be installed on any Windows® PC and uses the PC’s network interface card to generate and receive ED-137 traffic. The software can generate more than 500 simultaneous calls, emulate hundreds of user agents and full functionality of the Controller Working Position (CWP) entities in Ground-to-Ground telephone calls.

Recent Enhancements

  • Enhanced to support Python Client Application Programming Interfaces (APIs) to operate in Command Line Interface (CLI) mode
  • Support Meet Me Conference call feature
  • Provides option to insert SIP headers during run-time in CLI mode
  • Validated against VOTER_4.1.30.3 for VCS-Telephony-Interface-Test Cases

Key Features

  • Portable and easy to use during field installation, system configuration, and commissioning
  • Supported call types include
    • Instantaneous Access
    • Priority Direct/Indirect Access
    • Routine Tactical Direct/Indirect Access
    • Routine Strategic Direct/Indirect Access
    • Routine General Purpose Direct/Indirect Access
    • Position Monitoring (Combined A-G and G-G, A-G only, and G-G only) calls
    • Addendum 2: FAA Legacy Telephone Interworking
    • Addendum 4: Override Call
    • Addendum 5: Voice Call
  • Emulate different call scenarios like Call Hold, Call Transfer (attended and unattended), Call Pick-up, Preset Conference call, Broadcast Conference call, etc.
  • Easy to understand Call Flow Graphs of SIP message exchanges and displays message contents (SIP headers and SDP attributes)
  • Allow call rejection using SIP error codes (4xx, 5xx, 6xx)
  • RTP Traffic can be impaired by packet loss, latency, duplicated and out-of-sequence packets
  • Allows the user to define DSCP (Differentiated Service Code Point) values for signaling and voice traffic
  • Support complete customization of SDP and SIP headers, call flow, and messages
  • Tests can be run sequentially, randomly, or simultaneously for multiple iterations
  • Run sets of test cases automatically at a predefined time with a Scheduler feature
  • Support IP address spoofing for each endpoint to generate calls using a unique IP address from a single system
  • Support audio codecs such as G711 U-Law, A-Law, and G729

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