SIPSorcery
Browser-Phone
Our great sponsors
| SIPSorcery | Browser-Phone | |
|---|---|---|
| 9 | 4 | |
| 1,300 | 428 | |
| 3.2% | - | |
| 8.3 | 5.1 | |
| 1 day ago | 30 days ago | |
| C# | JavaScript | |
| GNU General Public License v3.0 or later | GNU Affero General Public License v3.0 |
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
SIPSorcery
- VoRS: Vo(IP) Simple Alternative to Mumble
-
Pure C WebRTC
I am really excited about https://github.com/sepfy/libpeer. It has examples ready for ESP32 etc....
When working on KVS I wasn't familiar with the embedded space at all. I saw 'heavyweight' embedded where you were running on Linux. Then you had RTOS/No OS at all. I wasn't prepared for these devices at all. If we can make WebRTC work in the embedded space I think it will really accelerate what developers are able to build!
Remotely driven cars, security cameras, robots in hospitals that bring iPads to infectious patients etc... Creative people are building amazing things. The WebRTC/video space needs to work harder and support them :)
-----
I love how diverse the WebRTC space is now. Outside of this implementation you have plenty of other options!
* https://github.com/shinyoshiaki/werift-webrtc (Typescript)
* https://github.com/pion/webrtc (Golang)
* https://github.com/webrtc-rs/webrtc (Rust)
* https://github.com/algesten/str0m (Rust)
* hhttps://github.com/sepfy/libpeer (C/Embedded)
* https://webrtc.googlesource.com/src/ (C++)
* https://github.com/sipsorcery-org/sipsorcery (C#)
* https://github.com/paullouisageneau/libdatachannel (C++)
* https://github.com/elixir-webrtc (Elixir)
* https://github.com/aiortc/aiortc (Python)
* GStreamer’s webrtcbin (C)
See https://github.com/sipsorcery/webrtc-echoes for examples of some running against each other.
- WebRTC for the Curious
-
Building WebRTC server implementation for Erlang
This is not true, there are actually multiple WebRTC implementations in different languages besides the reference library: aiortc (python), libdatachannel (C++), sipsorcery (C#),webrtc-rs (rust), werift (Typescript), and Amazon Kinesis (C)
-
Why WebRTC is not able to capture screen when the System is signed out or logged out?
I checked another example in Sipsorcery and it works. In this example it show a basic test pattern. For screen sharing Sipsorcery uses FFMPEG. I think FFMPEG has the restriction to capture the screen when the system is signed out.
-
Need help to connect to WebSocket signaling for WebRTC, C#
I was going through FFmpegFileAndDevicesTest, it is created using Sipsorcery library. In this code a websocket server is created and the PeerConnection is added to the websocket. Is there any way to connect to an external websocket server instead of creating one. The code is
- Need some help on C# WebRTC Signaling using WebSocket
- Show HN: WebRTC-Echoes: Interop for C#, C++, Python, TypeScript, Go and Servers
Browser-Phone
-
Building a Personal VoIP System
Which clients do you use? And how do you connect to a SIP trunk / DID without Asterisk?
I use WebRTC with Asterisk, and Browser Phone for the client (https://github.com/InnovateAsterisk/Browser-Phone). I don't use it much, but good enough for the rare times I have to use the phone.
-
Create a javascript client over Astersik's webrtc feature
Also have a look on this browser phone written using sipjs library for any help - https://github.com/InnovateAsterisk/Browser-Phone
-
Chat solution with SIP integration
If you want to go webrtc route you can check https://github.com/InnovateAsterisk/Browser-Phone which works with FS too.
-
Cant Connect WebRTC from Issabel to External SIPPhone.
with this SIPPhone in WebRTC( https://github.com/InnovateAsterisk/Browser-Phone.git)
What are some alternatives?
Pion WebRTC - Pure Go implementation of the WebRTC API
docker-asterisk - Docker image providing Asterisk PBX
libdatachannel - C/C++ WebRTC network library featuring Data Channels, Media Transport, and WebSockets
SIP.js - A simple, intuitive, and powerful JavaScript signaling library
aiortc - WebRTC and ORTC implementation for Python using asyncio
mirotalksfu - 🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
janus-gateway - Janus WebRTC Server
Asterisk-examples - PABX examples used in "Fixed and Mobile Telephone Systems" VET classes.
webrtc-echoes - Simple useful interoperability tests for WebRTC libraries. If you are a WebRTC library developer we'd love to include you!
slipstream - NAT Slipstreaming allows an attacker to remotely access any TCP/UDP services bound to a victim machine, bypassing the victim’s NAT/firewall, just by anyone on the victim's network visiting a website
webrtc - A pure Rust implementation of WebRTC
RTC-Call-Monitor - Voice/video call detection by monitoring of UDP packet rate with notification via webhooks