imankulov / network-emulatorLinks
Utility to test how network losses affects speech quality in VoIP-based applications
☆24Updated 12 years ago
Alternatives and similar repositories for network-emulator
Users that are interested in network-emulator are comparing it to the libraries listed below
Sorting:
- Linphone.org mirror for mediastreamer2 (git://git.linphone.org/mediastreamer2.git)☆137Updated this week
- Linphone.org mirror for ortp (git://git.linphone.org/ortp.git)☆163Updated this week
- Applying webrtc's acoustic echo cancellation (AEC) to audio files☆36Updated 9 years ago
- WebRTC AudioProc (AEC, VAD, NS...)☆105Updated 4 years ago
- voip packet-loss concealment algorithm derived from WebRTC neteq module☆30Updated 3 years ago
- codec for audio in G72X, G711,G723 G726 G729 and encode or decode them from PCM☆40Updated 9 years ago
- RTP Tools☆181Updated last year
- experimental version of G.729 codec for ARM devices☆27Updated 10 years ago
- webRTC aec模块 ，单独编译，相关头文件都已经整理☆44Updated 5 years ago
- RTP Audio Juicer☆25Updated 3 years ago
- Baresip WebRTC Demo - moved to baresip☆47Updated 3 years ago
- This is the Wrapper Library for WebRTC Voice Engine. Including Acoustic Echo Cancellation (AEC), Noise Suppression (NS), VAD (Voice Activ…☆125Updated 11 years ago
- generating problems on RTP streams : latency, delay, jitter☆16Updated 2 years ago
- Acoustic Echo Canceller for Mobile Module Port From WebRTC☆204Updated 5 months ago
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated last year
- Acoustic Echo Cancellation builtin WebRTC aec/aecm(mobile) module, speex 1.0/1.2.☆78Updated 7 years ago
- Sip Express Media Server☆176Updated this week
- PCM to G711 Fast Conversions☆68Updated 7 years ago
- PTLib 2.10.9 fork for GNU Gatekeeper and H323Plus☆24Updated last month
- ☆84Updated 6 years ago
- Comfort Noise Generator Module Port From WebRTC☆20Updated 6 years ago
- 此DEMO为webrtc中AEC3模块单独扣出,并使用远端参考信号和近端信号两个.wav文件进行测试，输出经过AEC3处理后的.wav音频文件。☆24Updated 2 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆68Updated this week
- Examples of SIP register UA with sofia-sip, pjsip, libeXosip and libre☆29Updated 7 years ago
- Voice Activity Detector Module Port From WebRTC☆179Updated 5 years ago
- GNU Gatekeeper - H.323 server for VoIP and videoconferencing☆82Updated last month
- SIP Session Border Controller Library☆26Updated 3 years ago
- 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
- AEC3 Extracted From WebRTC☆189Updated 3 years ago
- BFCP message/client/server libraries☆13Updated 6 years ago