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A SIP phone daemon with local speech and language models on Apple Silicon.

SIP to WebRTC bridge for LiveKit

A SIP/WebRTC voice agent

baresip library based SIP client for Android

SIP softphone for Mac

C++ implementation of SIP, ICE, TURN and related protocols.

Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.

OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms.

Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms, focusing on flexibility, security and scalability

Sip Express Media Server

Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.

Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of git://git.linphone.org/linphone-desktop.git

SIP softphone

Modern and flexible SIP/VoIP cli tool

纯协议层框架，屏蔽 SIP 协议细节。业务方实现 Listener 接口（推荐）或监听 Layer 1 协议事件 接收消息，通过 ClientCommandSender / ServerCommandSender 发送命令，不直接接触 JAIN-SIP。框架内置 GB28181-2016 + GB/T 28181-2022 协议实现，单 JVM 可同时启用平台服务端（gb28181-server）和设备客户端（gb28181-client），支持级联代理场景。

Voglander 是基于 Spring Boot 3.5 + Java 17 的企业级视频监控平台，支持 GB28181、JT/T 1078、ONVIF、SIP Gateway、ZLMediaKit、IPC/NVR 接入、实时点播、录像回放、流媒体转发、NAT 穿透和国标级联。

SIP Stack Rust library for building SIP applications

VoIP Stack for ESPHome and Home Assistant - local SIP phones, HA softphone/router, phonebook, softphone accounts and optional trunk calls

Open-source realtime voice agent server in Go with WebRTC (WHIP), barge-in, streaming STT/LLM/TTS pipelines, plugin system, multi-language SDKs, SIP telephony, ESP32 support & fully local mode.

Tiling that feels like it shipped with macOS. Works out of the Box, settings menu, in-app updates, native Desktops kept intact, Lua underneath. Accessibility only, never SIP.

Self-hosted API-first AI call center: AI talks, code controls the call. AI agents answer every call; humans take over without dropping it. Runs on FreeSWITCH with your SIP trunk and realtime model (OpenAI, Grok, Qwen, Doubao, or any compatible gateway).

One binary, one YAML file. A pure-Go Session Border Controller that keeps FreeSWITCH or Asterisk off the public internet: SIP proxy, RTP/SRTP relay, WebRTC gateway, built-in SIP security.

Remove unwanted Apple Intelligence models without disabling SIP. CLI, nix-darwin, and Home Manager support.