Description
Add to wishlistWith our module, you only need 5-10 mins to get peer-to-peer HD voice call feature into your app, eliminating a lot of hard works such as: NAT, NAT traversal, Echo cancellation, Noise reduction/suppression, SRTP, Jitter buffer,…
Our module focuses on solving voice processing such as: record audio, encoding, packaging and encrypting data in SRTP packet, transport data, NAT traversal, decoding… Call signaling is implemented by yourself such as: notify incoming call, notify ringing, notify answered, notify end call…
Features
– Excellent and stable voice quality
– Reduce latency and packet loss by peer-to-peer connection
– Contain various audio codecs: G711, iLBC, Speex, Opus, G722, GSM, …
– Noise reduction/suppression
– Echo cancellation
– Adaptive jitter buffer
– Written in C language to reduce memory usage and optimize for speed
– Voice stream operated through SDK are encrypted with SRTP