A modern CLI tool for crafting and sending SIP requests to test VoIP and WebRTC signaling servers.
sipexer is a command-line tool for crafting and sending SIP requests to test VoIP and WebRTC signaling servers. It allows developers to simulate SIP traffic, test server responses, and debug SIP routing with a flexible template system. The tool supports various SIP methods, authentication mechanisms, and transport protocols like UDP, TCP, TLS, and WebSocket.
Network engineers, VoIP developers, and QA testers who need to test SIP server functionality, signaling routing, or monitor SIP-based communication systems.
Developers choose sipexer for its flexibility in crafting custom SIP messages, support for modern protocols like WebSocket for WebRTC, and its lightweight, cross-platform design compared to heavier SIP softphones or testing suites.
Modern and flexible SIP/VoIP cli tool
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Uses Go's text/template to craft custom SIP requests, with fields settable via CLI or JSON, as detailed in the Message Template section.
Supports UDP, TCP, TLS, and WebSocket/WSS, enabling testing across modern VoIP and WebRTC environments, per the Features list.
Offers plain and HA1 password authentication with various hashing algorithms like MD5 and SHA256, detailed in the authentication features.
Includes variables for random numbers, timestamps, UUIDs, and environment variables, enhancing test customization as shown in Template Fields.
Requires knowledge of SIP protocol and Go template syntax to effectively use the template system, which can be complex for newcomers.
Focuses only on SIP signaling without media stream support, making it unsuitable for end-to-end call testing involving RTP.
Advanced usage necessitates creating and managing custom template files, adding overhead compared to more automated tools.