Show HN: Bring phone calls into the browser (sip-to-WebRTC)

• broadcast-box

  10 518 8.8 Go

  A broadcast, in a box.

  

> [1] https://github.com/Glimesh/broadcast-box

This is really interesting! Any latency benchmarks comparing it to gstreamer UDP, P2P primarily?

• example-webrtc-applications

  8 947 8.2 Go

  Examples of WebRTC applications that are large, or use 3rd party libraries

  

• WorkOS

  workos.com sponsored

  The modern identity platform for B2B SaaS. The APIs are flexible and easy-to-use, supporting authentication, user identity, and complex enterprise features like SSO and SCIM provisioning.

  

• webrtc-for-the-curious

  13 1,879 5.2 Python

  WebRTC for the Curious: Go beyond the APIs

  

• libpeer

  11 774 8.1 C

  WebRTC Library for IoT/Embedded Device using C

• str0m

  10 239 9.8 Rust

  A synchronous sans I/O WebRTC implementation in Rust.

• ex_webrtc

  1 215 8.8 Elixir

  An Elixir implementation of the W3C WebRTC API

  

• sip

  1 56 9.0 Go

  SIP to WebRTC bridge for LiveKit

  

Hi Sean! I'm guessing this is derives as part of the recent collaboration with LiveKit to add SIP support, right?

I had been looking at https://github.com/livekit/sip the other day, so it hasn't been much of a surprise to now see this here. Cool integration!

• InfluxDB

  www.influxdata.com sponsored

  Power Real-Time Data Analytics at Scale. Get real-time insights from all types of time series data with InfluxDB. Ingest, query, and analyze billions of data points in real-time with unbounded cardinality.

  

• obs-studio

  2,361 55,593 9.8 C

  OBS Studio - Free and open source software for live streaming and screen recording

  

I don't! But when adding WebRTC support to OBS I would see 120ms https://github.com/obsproject/obs-studio/pull/7926

This is me (in Ohio) going to a Digital Ocean Droplet (NYC) and back.

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