sipp

SIPp generates SIP traffic to test VoIP systems, simulating user agents (UAC and UAS) that establish and release calls using INVITE and BYE methods. It provides dynamic statistics display including call rate, round trip delay, and message statistics, with periodic CSV dumps. The tool supports TCP and UDP over multiple sockets or multiplexed with retransmission management and adjustable call rates.

Use cases include load testing SIP servers, validating call flows, and simulating high-volume traffic scenarios. It can read custom XML files for detailed call sequences and inject values from CSV files during tests. Features like RTP echo and 3PCC mode enable advanced testing of media streams and third-party call control.

SIPp operates in client or server modes with embedded scenarios or custom files, offering precise control over call rates, timeouts, and behaviors. It handles authentication, retransmissions, and unexpected messages with configurable defaults.

SIPp uses embedded or custom XML scenarios to define SIP call flows, sending messages like INVITE, ACK, and BYE over UDP, TCP, or TLS. It manages multiple sockets per call or IP, with retransmission handling and dynamic rate adjustment. Statistics are updated in real-time on screen or logged to CSV, tracking calls, RT delays, timeouts, and retransmissions. RTP echo functionality repeats received UDP packets on specified ports, supporting media testing. Global timers, injection files, and behaviors control call lifecycle and error handling.