Information Technology

GL Enhances MAPS™ SIP to Support Instant Messaging Using MSRP for Next Generation 911 (NG9-1-1)

09 June 2018

General NG9-1-1 VoIP architecture. Source: GL CommunicationsGeneral NG9-1-1 VoIP architecture. Source: GL Communications

GL Communications Inc., a provider of telecom test and measurement solutions, has unveiled its enhanced SIP Protocol Emulator, referred to as MAPS™ SIP Protocol Emulator, to help users generate and receive SIP. Public safety communication service providers can deploy this test tool for testing NG9-1-1 emergency services and components within the ESInet and for MSRP Protocol Emulation for Session Based Instant Messaging implementation.

Speaking to the press, Mr. Vijay Kulkarni, CEO of GL Communications Inc., said, “NG9-1-1 networks based on NENA i3 standards define the network transformation route to receive and respond to NG9-1-1 emergency requests via multiple media sources such as SMS, video, email, and/or instant messaging (IM).”

He further added, “Major wireless carriers have committed to providing text-to-911 services to end users. In addition to these services that are primarily dependent on carriers, there are many OTT (Over the Top) instant messaging apps which connect users with the NG9-1-1 system directly.”

GL’s enhanced MAPS™ SIP with MSRP support for Instant Messaging allows SIP vendors, wireless carriers, NG9-1-1 service providers and emergency communications centers to test IP applications for satisfactory working of NG9-1-1 services prior to deployment.

Important Features of MAPS™ SIP

  • Supports UDP, TCP and TLS transport types
  • High-Density version capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls/platform)
  • MAPS™ CLI interface based on a client-server model allows users to control all features of MAPS™ through APIs (TCL, Python, VBScript and Java Client)
  • Supports almost all industry standard codec types - G.711 (mu-Law and A-Law), G.722, G.729, G.726, GSM, AMR, EVRC, SMV, iLBC, SPEEX, EVS, OPUS and more (*AMR and EVRC variants require additional licenses)
  • User-defined voice quality statistics for received RTP Traffic can be calculated and updated periodically during run-time to a CSV file
  • Supports 64-bit RTP core to enhance performance — handles increased call rate of up to 3,000 calls with high volume traffic
  • Supports both RTP G.711 Pass Through Fax Simulation and T.38 Fax Simulation over UDPTL
  • Bulk Video Call Generation supported with H.264 and H.263 video codecs
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