TMCnet News
GL Enhances VoIP Testing Tool By Erik Linask Associate Editor, Internet Telephony magazine GL Communications has announced enhancements to PacketGen, the company’s software-based VoIP testing tool. In addition to network and equipment analysis, the tool is designed for stress testing and real-time individual and bulk call generation, including SIP and RTP/RTCP traffic. The latest additions to PacketGen increase the potential applications of the software.
With CLI (Command Line Interface), PacketGen’s capabilities can now be incorporated into and automated test environment, allowing users to create their own scripts using simple text commands and run them at the Windows command line, a feature that is designed to replace the manual GUI. CLI also allows control of multiple SIP cores, as does the GUI. CLI also eases integration of PacketGen in to other third-party applications, even further increasing its flexibility. For network-wide voice quality testing, once users have configured PacketGen for call generation across their networks, the software can automatically generate and receive calls using the RTP traffic-scripting feature and synchronous voice file transmission and reception. In conjunction with GL’s VQT software, users can actively measure and monitor network voice traffic.
PacketGen’s new scripting feature allows a new way to perform RTP traffic actions, providing increased control and flexibility. Commands are available to perform all GUI functions. Calls can be individually configured for scripts, or the same script can be used across several calls, and users can load scripts and view script progress in real time. PacketGen also now includes an RTP Script Editor, which allows easy building of RETP scripts via a GUI application and point and click setup.
The latest version of PacketGen also includes a jitter buffer in the receive direction to handle jitter caused by transmission over a VoIP network. This is a global run time control and applies to all RTP streams. Static and dynamic jitter buffers are user configurable, including length and an on/off feature.
Release 3.0 also incorporates a new user-configurable jitter buffer, a new G.729B audio codec, packetization time configuration for RTP traffic, user-defined payload types, an audio stream utility, auto-refreshing SIP registrar server registration, and more.
Come see all the products and solutions you need to properly monitor and test your network’s VoIP traffic at INTERNET TELEPHONY Conference & EXPO in San Diego October 10-13. At ITEXPO, you will have the chance to meet face to face with more than 150 vendors who can help achieve all your IP Communications goals, and you will have the opportunity to sit in on countless conference tracks and keynote presentations, ranging from security to network migration to hosting opportunities and much, much more.
Erik Linask is Associate Editor of INTERNET TELEPHONY. Most recently, he was Managing Editor at Global Custodian, an international securities services publication. To see more of his articles, please visit Erik Linask’s columnist page.
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