Virtual PBX Featured Article

The GL SIP Protocol Emulator Will Improve Virtual PBX Performance

January 13, 2017

By Frank Griffin - Contributing Writer

The infrastructure needed to deliver IP communications has to be monitored at all times to ensure the network is functioning at optimal levels. This is more important than ever because more organizations are using VoIP and many of the features it offers, including hosted or virtual PBX. Having the right testing and simulation equipment can improve the quality of service (QoS) and provide better service level agreements (SLAs). The announcement of GL Communications' upgrade of its Message Automation & Protocol Simulation (MAPS) SIP testing is going to make it possible for operators to simulate any interface in a SIP network and perform protocol conformance testing for SIP protocol implementations.

The MAPS platform can simulate User Agent Client- UAC, User Agent Server-UAS, proxy, redirect, registrar and registrant servers for SIP testing. It is available as MAPS SIP Protocol Test Tool and MAPS SIP Conformance Test Suite to simulate any interface in SIP protocol implementations.

In accordance with the SIP specification of ETSI standard, the Conformance Scripts have more than 300 test cases that include general messaging and call flow scenarios for multimedia call session setup and control over IP networks. These test cases verify conformance of actions, including registration, call control, proxies and redirect servers as well as the reporting of logging and pass/fail results and security protocols.

Karthik Ramalingam, Senior Manager for Product Development of GL Communications, said, “Secure Real-time Transport Protocol (or SRTP) can provide encryption, message authentication, and replay protection to the RTP/RTCP traffic. SRTP traffic is initialized over TLS (Transport Layer Security) / Secure Sockets Layer (SSL) network protocol (OpenSSL) with a Certificate and Key. SRTP encrypts the actual media portion of the calls preventing eavesdropping and tampering.”

According to GL Communications, MAPS can be used to simulate any interface of the VoIP network, which give CSPs a quality control measure when all of the features of a virtual PBX solution are being tested. Ramalingam said the 64 bit RTP Core(PKS102), MAPS SIP can simulate up to 3,000 simultaneous calls with digit, voice file, single/ dual tones, FAX, IVR, video and voice RTP traffic acting as more than one SIP entity at a time and generating any SIP message in VoIP network.

Some of the key features of the enhanced MAPS SIP are generating valid and invalid messages, supporting complete customization of SIP headers, call flow, and messages, and SIP message templates that simplify the customization of protocol fields so that it can be accessed from the different protocol fields from the scripts.

Additional features include IPv4 /IPv6 and transport over UDP and TCP support and conference (third-party added), attended call transfer, and call forwarding, which are features that are essential to provide quality virtual PBX services.

Edited by Alicia Young



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