GL Announces Enhanced SIP Protocol Emulator
Gaithersburg, MD, January 12, 2017 --(PR.com)-- GL Communications Inc., announced today the release of its enhanced MAPS™ SIP Protocol Emulator that help user generate and receive SIP signalling with traffic.
Speaking to the press, Mr. Karthik Ramalingam, a Senior Manager for Product Development of the company said, “GL's Message Automation & Protocol Simulation (MAPS™) designed for SIP testing can simulate User Agents (User Agent Client- UAC, User Agent Server-UAS), Proxy, Redirect, Registrar and Registrant servers. This test tool/traffic generator can be used to simulate any interface in a SIP network and perform protocol conformance testing (SIP protocol implementations).
MAPS™ can be used to simulate any interface of the VoIP network. Single MAPS™ instance can act as more than one SIP entity at a time and can generate any SIP message on wire in VoIP network and hence equipment needed to test reduced.”
He added, “The MAPS™ SIP Conformance Scripts is designed with 300+ test cases, as per SIP specification of ETSI TS 102-027-2 v4.1.1 (2006-07) standard. Test cases include general messaging and call flow scenarios for multimedia call session setup and control over IP networks. Logging and pass/fail results are also reported. Test cases verify conformance of actions such as registration, call control, proxies and redirect servers.
“GL’s MAPS™ SIP is also available in High Density version (requires a special purpose network appliance and PKS109 RTP HD licenses). This is capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls/platform).”
“While explaining the enhanced features, Mr. Ramalingam further said, “With the purchase of 64 bit RTP Core (PKS102), MAPS™ SIP can simulate up to 3000 simultaneous calls with various RTP traffic such as, digits, voice file, single/ dual tones, FAX, IVR, Video and Voice Quality testing over IP networks.
“Secure Real-time Transport Protocol (SRTP) can provide encryption, message authentication, and replay protection to the RTP/RTCP traffic. SRTP is initialized over Transport Layer Security (TLS) / SSL (OpenSSL) with a Certificate and Key. SRTP encrypts the actual media portion of the calls preventing eavesdropping and tampering.
“The CLI interface developed for MAPS™ allows users to control all features of MAPS™ through Python and Java APIs. The Python and Java API are an object-oriented coding paradigm, designed to give the user simple and intuitive containers for their messages, calls and regressions. The APIs is divided into 'High' and 'Low' level function calls. Both High and Low, can extract complete decode of all messages in a call along with the timestamp of when that message was transmitted or received.”
Some of the Important Features
· Generates and processes SIP valid and invalid messages
· Supports complete customization of SIP headers, call flow, and messages
· Each SIP message template facilitates customization of the protocol fields and access to the various protocol fields from the scripts.
· Supports IPv4 /IPv6 and transport over UDP and TCP, and TLS for secure transport
· Handles Retransmissions of messages with specific interval
· Scripted call generation and call reception.
· Supports conference (third-party added), attended call transfer, and call forwarding.
· Supports transmission and detection of various RTP traffic such as, digits, voice file, single tone, dual tones, IVR, FAX, and Video in IP networks.
· 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, and more. *AMR and EVRC variants require additional licenses.
· Supports both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211) simulation over IP.
· Test IVR, and Instant messaging features.
· Capability to generate more than 500 simultaneous video calls on Core i7 systems and more
About GL Communications Inc
GL Communications Inc, is a global provider of test and measurement solutions and has over the years worked with major telecom equipment vendors, service providers, and system integrators to meet the testing requirements arising at various stages of telecom products development life cycle.
GL offers a broad set of test solutions that help perform all types of testing on networks, from initial system design, to fine-tuning, troubleshooting, live deployment, and monitoring. The products are widely used to verify and ensure 'quality and reliability' of Wireless (4G LTE, 3G, 2G), SONET/SDH, IP, TDM, and PSTN networks.
GL core product development is backed by a strong team of R&D experts to match evolving market and technical challenges in a most cost-effective and innovative way.
Contact:
Shelley Sharma
Phone: 301-670-4784
E-mail: info at gl.com
Speaking to the press, Mr. Karthik Ramalingam, a Senior Manager for Product Development of the company said, “GL's Message Automation & Protocol Simulation (MAPS™) designed for SIP testing can simulate User Agents (User Agent Client- UAC, User Agent Server-UAS), Proxy, Redirect, Registrar and Registrant servers. This test tool/traffic generator can be used to simulate any interface in a SIP network and perform protocol conformance testing (SIP protocol implementations).
MAPS™ can be used to simulate any interface of the VoIP network. Single MAPS™ instance can act as more than one SIP entity at a time and can generate any SIP message on wire in VoIP network and hence equipment needed to test reduced.”
He added, “The MAPS™ SIP Conformance Scripts is designed with 300+ test cases, as per SIP specification of ETSI TS 102-027-2 v4.1.1 (2006-07) standard. Test cases include general messaging and call flow scenarios for multimedia call session setup and control over IP networks. Logging and pass/fail results are also reported. Test cases verify conformance of actions such as registration, call control, proxies and redirect servers.
“GL’s MAPS™ SIP is also available in High Density version (requires a special purpose network appliance and PKS109 RTP HD licenses). This is capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls/platform).”
“While explaining the enhanced features, Mr. Ramalingam further said, “With the purchase of 64 bit RTP Core (PKS102), MAPS™ SIP can simulate up to 3000 simultaneous calls with various RTP traffic such as, digits, voice file, single/ dual tones, FAX, IVR, Video and Voice Quality testing over IP networks.
“Secure Real-time Transport Protocol (SRTP) can provide encryption, message authentication, and replay protection to the RTP/RTCP traffic. SRTP is initialized over Transport Layer Security (TLS) / SSL (OpenSSL) with a Certificate and Key. SRTP encrypts the actual media portion of the calls preventing eavesdropping and tampering.
“The CLI interface developed for MAPS™ allows users to control all features of MAPS™ through Python and Java APIs. The Python and Java API are an object-oriented coding paradigm, designed to give the user simple and intuitive containers for their messages, calls and regressions. The APIs is divided into 'High' and 'Low' level function calls. Both High and Low, can extract complete decode of all messages in a call along with the timestamp of when that message was transmitted or received.”
Some of the Important Features
· Generates and processes SIP valid and invalid messages
· Supports complete customization of SIP headers, call flow, and messages
· Each SIP message template facilitates customization of the protocol fields and access to the various protocol fields from the scripts.
· Supports IPv4 /IPv6 and transport over UDP and TCP, and TLS for secure transport
· Handles Retransmissions of messages with specific interval
· Scripted call generation and call reception.
· Supports conference (third-party added), attended call transfer, and call forwarding.
· Supports transmission and detection of various RTP traffic such as, digits, voice file, single tone, dual tones, IVR, FAX, and Video in IP networks.
· 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, and more. *AMR and EVRC variants require additional licenses.
· Supports both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211) simulation over IP.
· Test IVR, and Instant messaging features.
· Capability to generate more than 500 simultaneous video calls on Core i7 systems and more
About GL Communications Inc
GL Communications Inc, is a global provider of test and measurement solutions and has over the years worked with major telecom equipment vendors, service providers, and system integrators to meet the testing requirements arising at various stages of telecom products development life cycle.
GL offers a broad set of test solutions that help perform all types of testing on networks, from initial system design, to fine-tuning, troubleshooting, live deployment, and monitoring. The products are widely used to verify and ensure 'quality and reliability' of Wireless (4G LTE, 3G, 2G), SONET/SDH, IP, TDM, and PSTN networks.
GL core product development is backed by a strong team of R&D experts to match evolving market and technical challenges in a most cost-effective and innovative way.
Contact:
Shelley Sharma
Phone: 301-670-4784
E-mail: info at gl.com
Contact
GL Communications Inc
Mr. Shelley Sharma
301-670-4784
www.gl.com
Mr. Shelley Sharma
301-670-4784
www.gl.com
Contact
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