IP/VoIP Protocol Simulators

Core Network Simulators

MAPS™ ED-137 Telephone Emulator

Ground-to-Ground Calls Simulation

  Brochure

Overview

Recent advances in Air Traffic Management over IP network has opened both opportunities for providing better services, and challenges to ensure reliability, and performance. Among many other solutions for testing Air Traffic Control network, GL offers MAPS™ ED-137 Telephone (PKS119) which can simulate Ground-Ground calls. MAPS™ ED-137 supports VoIP implementation as per ED-137B volume 2 and ED-137C volume 2- of EUROCAE (European Organization for Civil Aviation Equipment) standards.

MAPS™ ED-137 Telephone simulates the functions of Controller Working Position (CWP) in Ground-to-Ground telephone calls. The product supports transmission and detection of various RTP audio traffic such as real-time audio, voice file, digits, single tone and dual tones. The MAPS™ profile editor feature allows users to easily configure multiple CWPs profiles, allowing to simulate multiple CWP entities.


MAPS™ ED-137 Telephone now supports below addendums optionally.

  • Addendum 2: FAA Legacy Telephone Interworking
  • Addendum 4: Override Call
  • Addendum 5: Voice Call

Application is also enhanced to support additional call scenarios such as Call Pick-up, Preset Conference call and Broadcast conference call.

GL tools for signaling emulation and voice quality testing offer an end-to-end test solution for testing connections from the radio interfaces to the Controller Working Position (CWP) and recording voice communications in the ATM network. GL’s Air Traffic Management Solution also includes MAPS™ ED-137 Recorder Emulator and MAPS™ ED-137 Radio Emulator.

GL also provides a GUI based PacketScan - All IP Traffic Analyzer for online capture and decode of the signaling in real time both during tests and as a stand-alone tracer for live systems.

Main Features

Signaling and Traffic Simulation

  • Emulates ED-137/2B Telephone interface at CWP endpoints
  • Enhanced to support multiple CWP endpoints simulation
  • Portable, easy to configure and use during in-the-field installation, system configuration/ test 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, and Position Monitoring (Combined A/G and G/G, A/G only, and G/G only) Call
  • Supports SIP Headers defined in ED-137/2B
  • Depicts easy to understand Call Flow Graphs of SIP message exchanges and Displays Message contents (SIP headers and SDP attributes)
  • Allows call rejection through use of SIP response codes (4xx, 5xx, 6xx)
  • Supports multiple Profiles (Users/End points) from single node
  • Supports hundreds of simultaneous calls and load generation can be automated completely along with traffic
  • Allows to define DSCP (Differentiated Service Code Point) values for signaling and voice traffic
  • Supports complete customization of SDP and SIP headers, call flow, and messages
  • Supports both UDP and TCP (IPv4 and IPv6)
  • Handles Re-transmissions of messages with specific interval
  • Supports User authentication with Proxy, Registrar servers
  • Supports IP address spoofing for each endpoint to generate call using different IP address from a single system
  • Supports OPTIONS PING feature used to check the connection status
  • Supports additional call scenarios like Call Hold, Attended Call Transfer, Unattended Call Transfer, Call Pick-Up, Preset Conference, Broadcast Conference and more
  • Supports Addendum 2 (FAA Legacy Telephone Interworking), Addendum 4 (Override Call), and Addendum 5 (Voice Call) telephone specifications
  • Supports additional call features Call Pickup, Preset Conference call, and Broadcast Conference call

Traffic

  • Supports various traffic actions on the call such as Playback to Speaker, Send and Record audio file, Generate and detect inband digits, single tone and dual tone
  • Supports ED-137 defined codecs - G.711 (mu-Law and A-Law) and G.729
  • Supports User-defined and automated traffic actions on the call
  • Applies impairments to the traffic such as Packet Loss, Latency, Duplicate and Out of sequence
  • Provides aggregated voice quality statistics such as MOS/R-Factor, Packet Loss, Duplicate and out of sequence packets

GUI Features

  • Call Generation and Call Reception Window includes button options to apply Events on an ongoing call
  • Provides call statistics such as Active, Completed, Passed, Failed and Calls per second
  • Provides Event logs, Captured errors and Error events
  • Automation, Remote access, and Schedulers to run tests 24/7
  • Supported on Windows® 7, 8 or higher version operating systems
  • Supports 64-bit version to enhance signaling performance

CLI

  • Supports Client-Server functionality requires additional license; clients supported are TCL, Python, VBScript, Java, and .Net

Applications

  • Fully integrated, complete test environment for Air Traffic Management
  • Supports testing CWP, VCS, GRS (or RMG), and VRS
  • Handles strict routing and loose routing, when requests are routed through proxies

MAPS™ ED-137 Telephone Simulation

Typical SIP Procedure


A typical ED-137 Telephone call between CWP entities in the Ground-to-Ground call using MAPS™ ED-137 at one of the peer-end is as shown.

Ground-Ground Call Simulation (as per ED137_2B: Telephone)

MAPS™ ED-137 Telephone can be configured as CWP for placing and receiving calls, thus generating telephone calls.

Once call is established between the two terminals, transmission and detection of various RTP audio traffic such as real-time audio, voice file, digits, single tone and dual tones are also supported with additional licenses.

User-friendly GUI features allows users to manually start/stop traffic, impair the traffic (latency, packetloss, and packeteffect), transfer call, put active call on hold, and playback the call using Speaker On option.

Supported call types includes the following -

  • 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) Call

MAPS™ ED-137 Telephone (CWP)
– Routine Strategic Access Call Generation


MAPS™ ED-137 Telephone (CWP)
– Routine Strategic Access Call Reception
   

MAPS™ ED-137 Telephone Addendums

EUROCAE ED-137C Volume 2 specification describes basic requirements to establish, terminate, and modify speech media sessions of the Ground Telephone Service in an Air Traffic Services Ground Voice Network (AGVN). Along with the basic requirement, it also describes specific call functionalities through number of addendums. GL's MAPS™ ED-137 Telephone is now compliant with addendum 2, addendum 4 and addendum 5 specifications. Support for addendums are optional.


Addendum 2: FAA Legacy Telephone Interworking

Addendum 2 describes the provisions for the FAA Legacy Telephone Interworking, and translation between SIP endpoint(s) and an analog trunk interface to a legacy FAA VCS. The signaling translation between the SIP endpoints and the analog trunk is provided via a gateway. It details the interaction between the analog trunk and the SIP endpoint(s) for five FAA Legacy G/G call types. The supported call types are:

  • Legacy Dial-Selective DA/IDA Call
  • Legacy Selective Signaling DA/IDA Call
  • Legacy Non-Selective DA/IDA Call
  • Legacy Voice Call
  • Legacy Override (OVR) Call

Sample Call Simulation of Legacy Dial-Selective DA/IDA Call:

This call specifies the signaling interworking of a FAA Legacy Dial-Selective DA/IDA call between SIP endpoint and a Dial-Selective trunk interfacing to a Legacy VCS. To initiate a FAA Legacy Dial-Selective DA/IDA call, the calling party shall establish two-way RTP connectivity to the idle gateway.

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Legacy Dial-Selective DA/IDA Call

Addendum 4: Override Call

Addendum 4 describes the requirements for Override Calls. The Override Call Type is supplemental to the core ED-137C Volume 2 and meant to interact with other call types and features of the core volume; therefore, all applicable requirements established in the core ED-137C: Volume 2 apply to the Override Call Type.

The Override Call Type will provide transmit and receive audio from various call types and as such, will create scenarios where Loop Closure and Echo are to be detected and mitigated. The supported call types are:

  • Override Call
  • MD Override Call

Sample call simulation of Override call

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Override call

Sample call simulation of MD Override call

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MD Override call

Addendum 5: Voice Call

The Voice call type is supplemental to the core ED-137C Volume 2; therefore, all requirements established in the core ED-137C: Volume 2 apply to the Voice call type unless otherwise specified or superseded. The Voice call type is intended to act as a modified DA/IDA call that forms a new call type to satisfy implementation specific requirements. The Voice call type is initiated as a DA/IDA call.

However, upon initiation the Voice call provides an immediate Voice page from the calling to the called user, replacing ringing tones. The called party of a Voice call answers in the same way as a DA/IDA call and once active, the Voice call is maintained similarly as well. The supported call types are:

  • SD Voice Call
  • MD Voice Call

Sample call simulation of SD Voice call

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SD Voice call

Sample call simulation of MD Voice call

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MD Voice call

Additional Call Features

MAPS™ ED-137 Telephone supports below listed additional call features:

  • Call Pickup
  • Preset Conference
  • Broadcast Conference

Call Pickup

The Call Pickup services enables a user not involved in an early dialog to answer calls on behalf of other parties and may apply to Routine Direct/Indirect Access Calls. The party wishing to pick up the call sends a SUBSCRIBE to the remote entity to retrieve the dialog information. In addition, the party wishing to pick up an early dialog sends an INVITE with a Replaces Header to the calling party.

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Call Pickup

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Call Pickup notify

Preset Conference

A Preset Conference call is a Conference call type used to contact all other members assigned to a specific call. The initiating member of a Preset Conference call establishes contact with the call Focus with the intent of contacting the remaining members, via communication with the Focus. The dedicated script i.e. SipPresetCallControl.gls is used to simulate Preset Conference call.

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Preset Conference call

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Preset Conference call notify

Broadcast Conference

A Broadcast Conference call is an ad-hoc Conference call where a calling party can dynamically create a conference and add or remove other user agents. The execution of a conference is based on RFC 3891 (“Replaces Header”).

Basically, MAPS act as a SIP end-point and don’t have FOCUS capability. As a SIP user agent, we can establish a SIP session with FOCUS and SUBSCRIBE for a Conference event over the same dialog. User Event option is provided within the established dialog to indicate the FOCUS to Add and Remove conference parties using REFER method. On termination of the call we indicate the Un-Subscription of Conference event to FOCUS by setting Expires value as 0.

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Broadcast Conference call - Add Conference Party

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Broadcast Conference call - Remove Conference Party

Comparison

MAPS™ ED-137 Radio Simulation MAPS™ ED-137 Telephone Simulation
  • Emulates Radio interfaces at Controller Working Position (CWP) or Ground Radio Station (GRS) endpoints as per ED-137/1B Compliance
  • Emulates Air to Ground Calls
  • Supports generation of Real-time Session Supervision (R2S)-KeepAlive Packets and RTP Header Extensions to carry Radio signaling information as per ED-137/1B standard
  • Supports Normal/emergency radio sessions, all PTT Types and Radio call types
    • PTT Call types-  Normal PTT On, Coupling PTT On, Priority PTT On, and Emergency PTT On
    • Radio Call types – Radio Idle, Rx Only, Radio Tx Rx, and Coupling
  • Supported events - PTT on/off, Transmit CLIMAX Time Delay, send Re-Invite, receive traffic, impair traffic, and Play back the call on speaker
  • Option to define radio frequency, radio maintenance mode and permitted users list
  • Call graphs notifying all the above listed events with timestamp
  • Supports optional Climax Time Delay, RSSI Signal Quality Index and Radio Remote Control header extension types
  • CWP and GRS support WG67 KEY-IN event package (i.e. Subscribe/Notify of established Radio sessions at GRS)
  • Emulates Telephone interface at CWP endpoints as per ED-137/2B Compliance
  • Emulates Ground to Ground Calls
  • Supports all SIP Methods, Headers and Mandatory /Optional SDP attributes as per ED-137/1B
  • Supported call types include –
    • Instantaneous Access (IA)
    • Priority Direct/Indirect Access (DA/IDA)
    • Routine Tactical Direct/Indirect Access
    • Routine Strategic Direct/Indirect Access
    • Routine General Purpose Direct/Indirect Access
    • Position Monitoring Call (Combined A/G and G/G, A/G only, and G/G only)
    • Supported events -send Re-Invite, send traffic, receive traffic, put active call on Hold, transfer call, impair traffic, and Play back the call on speaker

Screenshots



Testbed Setup MAPS™ ED-137 – Telephone


Radio CWP Profiles


Radio GRS profiles


Telephone CWP Profiles

Call Graph


Call Statistics


Call Events Log

User defined Statistics

Resources


Webinars

VoIP ATM Test Solutions

Workshop 1 : MAPS™ ED-137 Radio Emulator

Workshop 2 : MAPS™ ED-137 Telephone Emulator

 

ED-137B Test Solutions

Testing VoIP Air-to-Ground Calls per ED-137C Radio

Enhanced Test Tools for ED-137B VoIP Air Traffic Control

ED-137 VoIP Emulation and Analysis for Air Traffic Management