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High-Density Call Generator for IP & Wireless Networks

MAPS™ High Density RTP Generator Software Ver 6.4.22 | Download Now!


Testing IP network performance with high volume calls is increasingly important to Wireless Carriers, Internet Service Providers, and Equipment vendors to ensure that quality of experience does not degrade with call intensity, volume, or protocol. Load and stress testing are important to verify stability of IP network and network elements pre- and post-deployment.

MAPS™ HD RTP (PKS109) is an advanced bulk call generator used to simulate high volume calls with traffic. It is available as special purpose rackmount network appliance with 4x1GigE NIC capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls/platform).


MAPS™ (Message Automation and Protocol Simulation) is GL’s standard framework for emulation of IP, TDM, and Wireless protocols.

MAPS™ HD RTP (PKS109) is an advanced bulk call generator used to simulate high volume calls with traffic. It is available as special purpose rackmount network appliance with 4x1GigE NIC capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls/platform).

The network appliance performs signaling and traffic generation for a vast array of communication protocols covering IP and Wireless networks. It supports simulation of SIP, IMS SIP UE, GSM A, BICC, MGCP, H.248/Megaco, and provides non-reference based voice quality using E-model (R-factor) and MOS with five mapping scales.

MAPS™ HD network appliance is designed to easily achieve up to 20,000 endpoints per appliance (5000 per port). Using a stack of multiple servers, a larger test system with 100K-200K calls (all controlled from a single Master Controller.) is achievable for enterprise to carrier grade testing.

The network appliance provides a modular and flexible solution to generate real voice calls using industry standard voice codecs such as G.711 A/µ-law, G.722, G.722.2 (AMR-WB), G.722.1, G.726, G.729A/B, GSM (EFR, FR and HR), AMR (Narrowband and Wideband), EVRC, EVRCB, EVRC-C, iLBC, Speex, SpeexWB, RFC 2833, and user-defined codecs for voice and tones.

GL also provides PacketScan™ HD (PKV120) for High Density IP Traffic Analysis w/ 4x1GigE Network Monitoring and PacketScan™ FB (PKV121), a File Based IP Traffic Analyzer Server for near real-time processing of traces.

MAPS™ HD Characteristics:
  • Transport over UDP and TCP, IPv4 and IPv6, and TLS for secure transport
  • Unique endpoint emulation using IP address, MAC address, and VLAN tagging
  • Easily achieve up to 20,000 endpoints per appliance (5000 per port)
  • Up to 350 calls per second (with RTP traffic)
  • Scales to around 100,000 to 200,000 endpoints with use of Master Controller for single point of control
  • Manage 10+ MAPS™ systems with single point of control from Master Controller
  • Simulate various traffic conditions to measure the performance and capability of an IP network element
  • Simulate complete protocol state machine to troubleshoot the network issues
  • Powerful Load Generation, Scheduler, and Command Line Interface (CLI) for automation, remote control, and load testing
  • Configurations, test scripts, and profiles can be saved and reused on a different systems
  • Easy-to-use MAPS™ scripting interface (UI) for test scripts to be created
  • User can modify any signalling messages, message parameters, and/or information elements.
  •  User can also add, modify or remove any optional parameter depending on specific feature of interest.
  • Packet level fault insertion can be performed by impairing any bit / byte of a message.
  • Real-time monitoring and reporting of registration and call statistics
  • Statistics can be viewed on any pair of endpoints
  • Export data to other applications for customized user report

Bulk Voice Traffic Simulation

RTP core can generate and receive voice traffic over IP networks and can work with applications such as GL’s MAPS™, VQuad™, and RTP ToolBox™ (PKS102). Transmit and Receive pre-recorded Voice Files in  wave, pcm, and GL's proprietary pre-compressed GLW files with a synchronous Tx/Rx option.  You can also directly send live voice using Talk using Microphone feature, and play the recorded voice files directly on to PC speakers. Some additional features that help in the voice traffic simulation are listed below-

    • Allows to specify a desired voice payload type to each codec for sending and receiving payload;
    • Sampling rate of the codec is displayed for the selected codec.
    • Comfort noise generation is supported for A-law, µ-law and G.726 codecs for sending and receiving payload.
    • Allows to set the buffer used for delayed packets that arrive at receiving end (both static and dynamic jitter buffers are supported)
    • Allows to set QoS (Type of Service) properties such as precedence, delay, throughput and reliability values to the outgoing stream
    • Comprehensive voice codec support -
      • G.711 (A-law / Mu-law - 64kbps)
      • G.711 App II (A-Law and Mu-Law with VAD Support)
      • G.722
      • G.722.1 (32 k and 24 k)
      • G.729, G.729B (8 kbps)
      • G.726 (5 bit 40kbps/4 bit 32kbps/3 bit 24kbps/2 bit 16 kbps)
      • G.726 (40/32/24/16 kbps with VAD)
      • GSM 6.10 (13.2 kbps)
      • GSM-HR (rate – 5.6kbps)
      • GSM-EFR (rate - 12.2kbps, packet time fixed at 20msec.)
      • SPEEX (Packet time fixed to 20msec)
      • SPEEX_WB (Packet time fixed to 20msec)
      • iLBC
      • iLBC_13_33
      • SMV (Modes - 0, 1, 2 and 3- Available if licenses are provided or owned, please call GL)
      • AMR (4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps) (optional codec)
      • AMR_WB
      • EVRC (Rates - 1/8, ½ and 1) , EVRC0 (optional codec)
      • EVRC_B (Rates - 1/8, ¼, ½, and 1), EVRCB0 (optional codec)
      • EVRC_C (optional codec)
      • H.263 video capture and conference capability
      • H.264 video compression codec

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Bulk RTP FAX Simulation (T.30 pass through and T.38 UDPTL)

GL offers a variety of test tools to perform FAX over IP (FoIP) simulation and monitoring. Fax simulator supports both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211). Almost all MAPS™ IP based simulation products supports FAX simulation using any of these two methods. Typical applications of our Fax Simulators include load testing of fax servers, qualification testing of T.38 Gateways, testing of ATAs (Analog Terminal Adapters), testing of internet-aware fax machines, and many more.

With respect to channel capacity RTP pass-through supports up to 120 Fax ports, whereas T.38 fax simulation over UDPTL supports unlimited channels, and constrained only by CPU capacity.

Almost all MAPS™ IP products support fax simulation – MAPS™ SIPMAPS™ SIP-IMAPS™ MEGACOMAPS™ BICC, MAPS™ GSM, and MAPS™ UMTS.

MAPS™ allows the user to initiate fax calls by sending call control messages using proper scripts and profiles. The profile allows necessary parameters of call control messages to be changed during runtime. The below figure depicts the T.30 fax call being generated using MAPS™ SIP.

GL’s RTP Fax Simulator simulates multiple fax calls over IP in T.30 pass through mode (using G.711 PCMU and PCMA). It can transmit pre-recorded Tiff image to DUT (Real-time Fax machine), receive Pass-Through fax from DUT, and record complete fax call messages as log file along with a Tiff image.

T.30 G.711 Pass through mode FAX simulation using MAPS™ SIP


MAPS™ SIP generates Re-Invite to switch from audio mode to image (FAX) mode. While the call in progress, the MAPS™ also provides events related to the progress of the call. After completion of the call, MAPS™ provide call quality statistics.

GL’s RTP Fax over UDPTL transport simulates multiple fax calls over IP using T38 protocol (compliant with ITU-T T.38 (03/2002)) up to maximum of 33.6 kbps speed.

The below figure depicts the T.38 fax call being generated using MAPS™ SIP.

T.38 Fax call in progress 


Bulk Video Traffic Simulation

MAPS™ SIP provides the Bulk Video Call Simulation capability using its pre-recorded video traces supporting codecs like H.264, H.263, & VP8. On a high-performance computing platform (core-i7), it is possible to generate more than 500 simultaneous video calls. With a High Density (MAPS™ HD)platform, it is possible to achieve much more capacity. H.263 provides video capture and video conference monitoring capability, while H.264 is an industry standard codec for video compression. H.264 codec offers better compression performance over previous standards.

Below figure depicts the bulk video call simulation and RTP video transmission in MAPS™ SIP. Observe the decode part of the INVITE message showing the information about the audio and video codecs used in the request.

Bulk Video Call Simulation in MAPS™ SIP


RTP/RTCP Packet Statistics

Statistics reports of RTP and RTCP packets transmitted on a session such as number of packets sent/received, dropped packets, out of sequence packets and more. Sender and receiver reports are also displayed using RTP/RTCP statistics applications.

RTP /RTCP Packet Statistics


Speech Quality Metrics (R Factor & MOS)

Quality metrics include various graphs for R-Factor and for MOS Factor. R Factor graph will display statistics such as, R-Listening, R-Conversational, R-G107 and R-Nom. MOS Factor graph will display statistics such as MOS CQ, MOS PQ and MOS Nom. The additional licensing (PKS108) RTP voice quality metrics for the received calls are calculated and are reported to MAPS™ application which are displayed including Listening and Conversational Quality MOS scores - MOS-LQ, MOS-CQ, and Listening and Conversational Quality R factors - R-LQ, R-CQ. Estimates are based on the ITU G.107 E Model.

A typical estimate of the MOS and R-Factor scores for each codec is available in

GL’s MAPS™ application provides a feature which helps users to customize the global statistics for RTP audio and video traffic. These global parameters are defined in the call generation scripts, which are calculated and updated periodically providing call quality metrics such as Listening MOS, Conversational MOS, PacketLoss, Discarded Packets, Out of Sequence Packets, Duplicate Packets, Delay and Jitter as shown in the figure below.



MAPS™ HD has many applications including:

  • Pre- and post-deployment testing of enterprise, ISP, and Carrier Networks
  • Quality of Experience (QoE) evaluation as a function of signaling or traffic load or protocol variation
  • Networks Applications for
  • IP, Mobile Broadband, SIP, IMS, Wireless
  • Quality of Service (QoS) for voice
  • End-to-End testing of complex networks like Next Generation Voice Networks, Unified Communications, and IMS & Contact Centers
  • Contact Center and Unified Communications
  • Interactive Voice Response System (IVR), Automatic Call Distributor (ACD), routing, Computer Telephony Integration (CTI), CRM, chat, presence, conferencing, speech, and agent desktop applications
  • Security
  • Denial of Service, Theft of Service, traffic floods, intrusion prevention, firewalls, intrusion detection systems, and security gateways
  • Network elements
  • Session border controllers, network border elements, application level gateways and deep packet inspection devices, and intrusion prevention/intrusion detection systems
  • Automated Tests
  • Load generation can test maximum active call handling capacity, busy hour call attempts, and calls attempted by the simulator at one shot (burst).
  • Stress testing can check for out-of-boundary conditions.
  • Regression tests can be automated and repeatable to ensure that all problems are corrected.

Buyer's Guide

Please Note: The XX in the Item No. refers to the hardware platform, listed at the bottom of the Buyer's Guide, which the software will be running on. Therefore, XX can either be ETA or EEA (Octal/Quad Boards), PTA or PEA (tProbe Units), UTA or UEA (USB Units), HUT or HUE (Universal Cards), and HDT or HDE (HD cards) depending upon the hardware.

Item No. Description
PKS109 MAPS™ High Density RTP Generator

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