imankulov / network-emulator
Utility to test how network losses affects speech quality in VoIP-based applications
☆24Updated 11 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:
- voip packet-loss concealment algorithm derived from WebRTC neteq module☆29Updated 3 years ago
- generating problems on RTP streams : latency, delay, jitter☆16Updated last year
- Applying webrtc's acoustic echo cancellation (AEC) to audio files☆36Updated 9 years ago
- Baresip WebRTC Demo - moved to baresip☆47Updated 2 years ago
- Acoustic Echo Cancellation builtin WebRTC aec/aecm(mobile) module, speex 1.0/1.2.☆76Updated 7 years ago
- ☆83Updated 6 years ago
- 回声消除☆23Updated 5 years ago
- WebRTC AudioProc (AEC, VAD, NS...)☆104Updated 4 years ago
- experimental version of G.729 codec for ARM devices☆26Updated 9 years ago
- codec for audio in G72X, G711,G723 G726 G729 and encode or decode them from PCM☆40Updated 9 years ago
- webRTC aec模块 ,单独编译,相关头文件都已经整理☆41Updated 5 years ago
- Comfort Noise Generator Module Port From WebRTC☆18Updated 6 years ago
- ☆25Updated 10 years ago
- SIP Express Media Server, very fast and flexible SIP media server☆65Updated last week
- 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。☆21Updated 2 years ago
- A package used to test webrtc apm functions, such as aec, ns☆16Updated 6 years ago
- Examples of SIP register UA with sofia-sip, pjsip, libeXosip and libre☆28Updated 7 years ago
- Unofficial mirror/fork of http://svn.pjsip.org/repos/pjproject/trunk/ — check the Wiki for more information.☆55Updated 10 years ago
- Linphone.org mirror for ortp (git://git.linphone.org/ortp.git)☆158Updated last week
- Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)☆41Updated last year
- Audio AEC AGC ANS(webrtc aec) process with a single lib, just 6 APIs, easy to use. 基于webrtc的极简音频3A处理上层封装,6个API,支持常见采样率、单双声道。☆37Updated last year
- Voice Activity Detector Module Port From WebRTC☆174Updated 4 years ago
- Audio Loudness Normalization Filter Port From FFmpeg☆11Updated 6 years ago
- RTP Audio Juicer☆24Updated 3 years ago
- Audio and video processing media library☆97Updated 3 years ago
- BFCP message/client/server libraries☆13Updated 5 years ago
- RTP Cluster is a front-end for multiple RTPproxies☆41Updated last year
- 单独移植编译webrtc的aec模块☆17Updated 6 years ago
- AEC3 Extracted From WebRTC☆174Updated 3 years ago
- The SIP Voice Quality Report Reaper sniffs RTCP and RTP packets and generates SIP PUBLISH messages with voice quality reports.☆17Updated 10 years ago