If you have just upgrading your VoIP infrastructure or purchased a high cost VoIP server/platform then it is the time to test it before you pay and before to launch your VoIP service in production. While in production, your server will receive similar traffic from attackers or while under high load legitimate traffic, so you might want to ensure that your service will behave correctly to avoid high cost incidents.
    VoIP network performance testing can mean the difference between a VoIP system working at a high level QoS and a weak system that runs so poorly customers could take their business elsewhere.  A virtual network test bed is particularly useful for taking risk out of both initial VoIP deployment and long-term VoIP ownership. Essentially, such a test bed enables application developers, QA specialists, network managers and other IT staff to observe and analyze the behavior of network applications in a lab environment that accurately emulates conditions on the current and/or planned production network.  Fast, accurate, and non-resource intensive voice quality testing is key to speeding network design and VoIP equipment deployment, as well as to meeting service quality objectives in operation.
    With this voipest (callgen and callterm) application you can easily stress test your VoIP network finding its weaknesses very quickly.
    The VoIP test application is able to generate and terminate a big amount of traffic having a multi-threaded VoIP engine written entirely in C++.
    The application can be used also by people without a deep understanding of the VoIP protocols. Just launch the application, start one of the tests and check if your VoIP server can survive.

Highlights:

• call generation module
• call termination module
• DoS and DDoS attack simulation
• up to 8000 simultaneous call from a single PC
• act as sip client or sip server
• register capability
• codec selection
• max simultaneous calls
• call recording option
• detailed log of outgoing and incoming calls
• detailed CDR records with caller id, called number, sip server, start-end, end-reason, etc
• programmed progression of calls
• fix called number or random called number or fix prefix and random suffix
• random delay between calls within the specified limits
• random connect time and call duration within the specified limits
• detailed rtp/media statistics (packet count, delay, packet loss, etc)
• presence and QOS emulations
• statistic output