Alternatives and similar repositories for MySipApp

Users that are interested in MySipApp are comparing it to the libraries listed below

• hujianhua888 / webrtc_vs2015
  ☆40Updated 6 years ago
• hanpfei / OpenRTCClient
  ☆75Updated 9 months ago
• zhanghuicuc / ffmpeg_video_combine
  本程序实现了对多路视频进行合并实现分屏效果,并且可以添加不同的视频滤镜。This program can combine multiple input videos and output one video which implementing a mutli-screen…
  ☆61Updated 2 years ago
• zhanghuicuc / WebRTC-VoiceEngine-Demo
  This project can be used to conduct audio call. 本程序可以实现语音通话。
  ☆48Updated 3 years ago
• ireader / 3rd
  ☆30Updated 2 years ago
• zhanghuicuc / WebRTC-VideoEngine-Demo
  This project can be used to conduct loopback video call, you can also choose to use OPENH264 or X264 as video codec besides VP8. 本程序可以实现环…
  ☆67Updated 3 years ago
• sxcong / pjsipvideo_demo
  sip video and voice client demo, receive rtsp media stream and push to other sip client. It use pjsip ,live555 and ffmpeg
  ☆79Updated 9 years ago
• ZhengfengRao / rtp2mp4
  recv rtp(h264+aac), save as mp4 file
  ☆117Updated 13 years ago
• xia-chu / ZLMediaPlayer
  一个简单的rtmp/rtsp播放器，支持windows/linux/macos
  ☆46Updated 5 years ago
• pcgpcgpcg / janus-gateway-android
  android rtc client for janus gateway
  ☆32Updated 6 years ago
• shenshuyu / es2ps
  transfer h264 ES stream to PS stream, written in C
  ☆44Updated 5 years ago
• xia-chu / ijkplayer
  基于ijkplayer的定制化修改，修复bug，开启FFmpeg硬件解码
  ☆29Updated 5 years ago
• 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
• str2num / libice
  libice是一个c/c++库，它实现了RFC5245规范定义的交互式连接建立(ICE)协议, 适用于实时通信领域，比如音视频实时通信。
  ☆119Updated 2 years ago
• 7956968 / LibGb28181PsMux
  PsMux for China GB28181
  ☆73Updated 3 years ago
• xia-chu / webrtc_apm
  webrtc中apm相关代码的提取，包括AEC/NS/AGC/VAD ，另外还包括mp3/aac编码器、SoundTouch
  ☆100Updated last year
• kingsunc / AVFileParse
  音视频文件格式解析
  ☆88Updated 7 years ago
• zhangpengyf / foxrtc
  media sdk based on webrtc
  ☆40Updated 2 years ago
• miaopei / ffmpeg-webrtc
  ffmpeg and webrtc note
  ☆15Updated 4 years ago
• webrtc-mirror / mirror
  WebRTC编译国内加速镜像
  ☆157Updated 4 years ago
• escoffier / sipsdk
  a gb28181 sip sdk based on resiprocate
  ☆30Updated 6 years ago
• zhiyong0804 / rtmp_pusher
  rtmp推送aac音频流
  ☆26Updated 5 years ago
• dengzikun / WebRTC-APM-for-Android
  ☆82Updated 5 years ago
• qwerty258 / librtp
  FFmpeg, live555 already support RTSP, SDP, RTP, RTCP.
  ☆16Updated last year
• zhoudd1 / GB28181Android
  GB28181 for Android, include RTP/PS/H264/G711
  ☆83Updated 5 years ago
• brookicv / FSplayer
  A simaple video player with FFmpeg and SDL2.0
  ☆92Updated 6 years ago
• lmshao / RTP
  Implementation of a RTP server that sends video stream (H.264/HEVC) using the Real-time Transport Protocol(RTP) based on Linux/MacOS. 一个基…
  ☆67Updated 6 years ago
• larkguo / SIP_UA
  SIP UserAgent(UAS and UAC) Sample
  ☆38Updated 8 years ago
• Pandalzm / mp4v2-h265
  ☆38Updated 4 years ago
• str2num / librtcbase
  一个移植于WebRtc项目的通用基础lib库
  ☆26Updated 6 years ago