Latest updates, releases, and announcements from the VoiceBlender project.
Benchmarking VoiceBlender's voice-agent pipeline: 1,200 concurrent SIP sessions with no capacity knee, p50 turn latency of 1,478 ms that does not move across a 48x range of load, and 0.0044 CPU cores per session.
Read more →VoiceBlender v0.12.0 brings SIPREC session recording on both sides, multi-stream SIP legs for live translation, Deepgram Flux turn detection with preflight TTS, live routing changes, and a Google Cloud Storage backend for recordings.
Read more →VoiceBlender now negotiates AMR-NB and AMR-WB natively through two pure-Go codec libraries - no cgo, RFC 4867 framing, within-2x throughput of the C reference, and an honest look at the AMR-WB encoder amplitude divergence and its mitigation.
Read more →VoiceBlender v0.8.0 lands native LiveKit integration as a first-class leg and inbound SIP REGISTER support over UDP, TCP and TLS for PBX-style deployments.
Read more →A roundup of recent VoiceBlender additions: WebSocket-as-a-leg, experimental MoQ over WebTransport, symmetric SIP NAT traversal, role-based audio routing, bridged rooms, Real-Time Text and full VSI control.
Read more →How a protocol I once dismissed became emergency-calling infrastructure - and how we added T.140 Real-Time Text to VoiceBlender as a Kamailio World Dangerous Demo.
Read more →VoiceBlender v0.5.0 introduces WhatsApp calling, configurable speech detection, call direction in ringing events, and bug fixes.
Read more →VoiceBlender v0.4.1 adds WebSocket control and events, configurable room mixing sample rate, Docker support, and more.
Read more →VoiceBlender v0.3.0 adds blind and attended call transfers, pause recordings, configurable jitter buffer, DTMF broadcast, and more.
Read more →VoiceBlender v0.2.0 brings Answering Machine Detection, Azure Speech Services support, improved speech detection, and more.
Read more →VoiceBlender is an open source SIP/WebRTC/AI voice framework.
Read more →