Alternatives and similar repositories for network-emulator

Users that are interested in network-emulator are comparing it to the libraries listed below

• jchavanton / disruptor
  generating problems on RTP streams : latency, delay, jitter
  ☆16Updated 2 years ago
• baresip / baresip-webrtc
  Baresip WebRTC Demo - moved to baresip
  ☆47Updated 3 years ago
• twstx1 / codec-for-audio-in-G72X-G711-G723-G726-G729
  codec for audio in G72X, G711,G723 G726 G729 and encode or decode them from PCM
  ☆40Updated 9 years ago
• TaoistKing / AudioPlc
  voip packet-loss concealment algorithm derived from WebRTC neteq module
  ☆30Updated 3 years ago
• BelledonneCommunications / bcg729
  Linphone.org mirror for bcg729 (git://git.linphone.org/bcg729.git)
  ☆129Updated last year
• DoubangoTelecom / g729
  experimental version of G.729 codec for ARM devices
  ☆26Updated 9 years ago
• DoubangoTelecom / webrtc-audioproc
  WebRTC AudioProc (AEC, VAD, NS...)
  ☆104Updated 4 years ago
• lschilli / wav-aec
  Applying webrtc's acoustic echo cancellation (AEC) to audio files
  ☆36Updated 9 years ago
• staskobzar / sip_stacks_examples
  Examples of SIP register UA with sofia-sip, pjsip, libeXosip and libre
  ☆29Updated 7 years ago
• irtlab / rtptools
  RTP Tools
  ☆179Updated last year
• BelledonneCommunications / mediastreamer2
  Linphone.org mirror for mediastreamer2 (git://git.linphone.org/mediastreamer2.git)
  ☆137Updated this week
• xshl5 / KOTI_AEC
  Acoustic Echo Cancellation builtin WebRTC aec/aecm(mobile) module, speex 1.0/1.2.
  ☆77Updated 7 years ago
• BelledonneCommunications / ortp
  Linphone.org mirror for ortp (git://git.linphone.org/ortp.git)
  ☆163Updated last week
• LuoZhongYao / webrtcaecm
  webRTC aec模块 ，单独编译，相关头文件都已经整理
  ☆44Updated 5 years ago
• sipwise / sems
  SIP Express Media Server, very fast and flexible SIP media server
  ☆68Updated this week
• dennisguse / ITU-T_pesq
  ☆84Updated 6 years ago
• joegen / oss_core
  SIP Session Border Controller Library
  ☆26Updated 3 years ago
• cpuimage / WebRTC_CNG
  Comfort Noise Generator Module Port From WebRTC
  ☆19Updated 6 years ago
• cpuimage / WebRTC_AECM
  Acoustic Echo Canceller for Mobile Module Port From WebRTC
  ☆200Updated 3 months ago
• cpuimage / FFmpeg_Loudness
  Audio Loudness Normalization Filter Port From FFmpeg
  ☆11Updated 6 years ago
• sippy / rtp_cluster
  RTP Cluster is a front-end for multiple RTPproxies
  ☆41Updated last year
• sipcapture / rtpbreakr
  RTP Audio Juicer
  ☆25Updated 3 years ago
• cpuimage / WebRTC_VAD
  Voice Activity Detector Module Port From WebRTC
  ☆179Updated 5 years ago
• xiph / webrtc-opus-ng
  ☆17Updated last year
• daimoc / Kurento-SIP-GW
  ☆35Updated 3 years ago
• jagger2048 / WebRtc_noise_suppression
  This is WebRtc noise suppression module demo.
  ☆103Updated 5 years ago
• willamowius / h323plus
  H323Plus - H.323 development framework
  ☆114Updated 2 months ago
• AGProjects / mediaproxy
  Media relay for RTP/RTCP and UDP streams
  ☆47Updated 4 months ago
• willamowius / gnugk
  GNU Gatekeeper - H.323 server for VoIP and videoconferencing
  ☆80Updated 2 months ago
• IBM / g729-codec-service
  Microservice for transcoding audio to/from G.729 Annex A/B and G.711 over a WebSocket connection.
  ☆14Updated 3 months ago