VoIP 开发
约 1 分钟阅读
JavaScript/WebRTC
-
onsip/sip.jsGitHubonsip/sip.js
- MIT, TypeScript
- WebRTC SIP 客户端库
- 支持:注册、音视频通话、IM、Presence、屏幕共享、DTMF
- sipjs.com
- Asterisk 配置指南
-
davehorton/drachtio-srfGitHubdavehorton/drachtio-srf
- MIT, JavaScript
- Node.js SIP Server 框架
- Drachtio Signaling Resource Framework
Go
- pion/webrtcGitHubpion/webrtcwebrtc/samples · WebRTC Web demos and samples · webrtc/apprtc · appr.tc · rtctunnel/rtctunnel · network tunnels over WebRTC笔记:WebRTC Awesome
- MIT, Go
- 纯 Go 实现的 WebRTC
- emiago/sipgoGitHubemiago/sipgo
- BSD-2, Go
- SIP 协议库
CPaaS 平台
- OnSIP
- SIP.js 创建者
- 提供 CPaaS API
关联信息
反向链接、本文链接的其他页面和外部资料。
参考资料
GitHub4 条
- davehorton/drachtio-srfgithub.com/davehorton/drachtio-srf
- emiago/sipgogithub.com/emiago/sipgo
- onsip/sip.jsgithub.com/onsip/sip.js
- pion/webrtcgithub.com/pion/webrtc
其他外链3 条
- developer.onsip.comdeveloper.onsip.com/
- sipjs.comsipjs.com/
- sipjs.com/guides/server-configuration/asterisksipjs.com/guides/server-configuration/asterisk