Alternatives and similar repositories for baresip-apps

Users that are interested in baresip-apps are comparing it to the libraries listed below

• juha-h / libwebrtc
  webrtc AEC module and its required dependencies module for android
  ☆19Updated last year
• sipXtapi / sipXtapi
  complete SIP signalling and RTP media service for rapid development of voice/video services and softphones
  ☆34Updated this week
• mingyuejingque / rtaudio_speex
  用rtaudio来采集、播放，并用speexdsp来做回声消除。
  ☆18Updated 7 years ago
• ouxianghui / mediasoup-server
  mediasoup c++ server (SFU)
  ☆14Updated last year
• juha-h / libbaresip-android
  libbaresip for Android
  ☆45Updated 2 weeks ago
• akumrao / mediaserver
  Live media streaming. High performance Http, secure websocket and webrtc server. Supports H264, Opus and Mp3.
  ☆16Updated last week
• wesley-fly / wtkrtc-sdk-api
  WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.
  ☆23Updated 6 years ago
• KingsleyYau / WebRTC-Server
  LiveStream gateway for WebRTC 流媒体网关服务器
  ☆25Updated last week
• baresip / re
  Generic library for real-time communications with async IO support
  ☆155Updated this week
• baresip / baresip-webrtc
  Baresip WebRTC Demo - moved to baresip
  ☆47Updated 3 years ago
• BarGabriel / SipServer
  A simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms.
  ☆95Updated last year
• wexiangis / wmix
  混音播放器，音频录、播音托管程序。支持wav、mp3、aac格式播放，rtp搭载pcma、aac发收测试，webrtc vad录音辅助(无说话时消音)、ns/nsx录、播音噪音抑制、aec/aecm回声消除、agc录音自动增益 等
  ☆16Updated 3 years ago
• willamowius / gnugk
  GNU Gatekeeper - H.323 server for VoIP and videoconferencing
  ☆81Updated 3 weeks ago
• yangkang2021 / libwebrtc
  webrtc各个版本在不同平台的库和demo，供上层用户使用。
  ☆13Updated 3 years ago
• yangkang2021 / mediasoup_server_cluster
  ☆17Updated 2 years ago
• Bepartofyou / janus-gateway-live
  RTMP edge speed with janus-gateway
  ☆47Updated 4 years ago
• X2Rtc / X2Rtc-Stream
  Open source, Open mind.
  ☆19Updated last year
• escoffier / sipsdk
  a gb28181 sip sdk based on resiprocate
  ☆30Updated 7 years ago
• mlfcjob / Rtsp2Rtmp
  A program transcode rtsp to rtmp with FFmpeg API.
  ☆17Updated 5 years ago
• pol51 / PJSIP-CMake
  PJSIP-CMake provides a CMake build system for PJSIP
  ☆23Updated last year
• webrtc-mirror / webrtc
  ☆21Updated this week
• fonnes / flexible_fec
  An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.
  ☆35Updated 5 years ago
• RSATom / janus-gstreamer-plugin
  [POC] GStreamer plugin for Janus Gateway
  ☆29Updated 2 years ago
• mail2chromium / Android-Native-Development-For-WebRTC
  A Complete Guide For WebRTC NDK Development for Audio Processing using Android Platform.
  ☆21Updated 4 years ago
• DrKLO / webrtc-google-h265
  ☆79Updated 2 years ago
• mqp / janus-helloworld-plugin
  Simplest possible plugin for Janus. Intended as a instructional sample for plugin builders.
  ☆33Updated 4 years ago
• signalwire / libks
  Foundational support for signalwire C products
  ☆45Updated last month
• cheungreed / StreamResolver
  1）完成H264、H265、G711和AAC几种音视频编码格式数据到RTP、PS、TS包的封装处理 2）解析获取RTP、PS、TS包中的裸数据
  ☆57Updated 5 years ago
• creytiv / rem
  Audio and video processing media library
  ☆97Updated 3 years ago
• byyam / mediasoup-go-worker
  implement of mediasoup-worker with golang
  ☆10Updated last year