Browser-Phone
SIP.js
| Browser-Phone | SIP.js | |
|---|---|---|
| 4 | 4 | |
| 433 | 1,822 | |
| - | - | |
| 4.8 | 0.0 | |
| 15 days ago | 2 months ago | |
| JavaScript | TypeScript | |
| GNU Affero General Public License v3.0 | MIT License |
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Browser-Phone
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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.
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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
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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.
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Cant Connect WebRTC from Issabel to External SIPPhone.
with this SIPPhone in WebRTC( https://github.com/InnovateAsterisk/Browser-Phone.git)
SIP.js
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Browser-to-Browser calling with SIP.js and Routr
SimplePhone is a phone built using SIP.js that runs as a Docker container. Please see the documentation at https://sipjs.com/ to develop your implementation.
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Create a javascript client over Astersik's webrtc feature
I found this library: https://github.com/onsip/SIP.js/releases
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[AskJS] Do you consider ESM to have more benefits than drawbacks in node?
Because it's implicit, CJS creates a bunch of resolution issues with Typescript, to the point where if you want to code for the browser, you can't easily can't and have to either recompile with something like webpack to rewrite the module resolution, or add .js to all your TS imports, which already looks weird (you'd think you're importing the .ts files). sip.js has that issue.
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Developers - Embed the Ability to Make a VOIP Call to My Asterisk in Web Page?
https://github.com/onsip/SIP.js or https://github.com/versatica/JsSIP combined with a SIP server like Asterisk/FreePBX or Freeswitch/FusionPBX
What are some alternatives?
docker-asterisk - Docker image providing Asterisk PBX
WebphoneLib - Easier web calling by providing a layer of abstraction around SIP.js
SIPSorcery - A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
JsSIP - JsSIP, the JavaScript SIP library
mirotalksfu - 🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
extless-loader - Loads Node ESM in a more compatible way by making extensions optional.
Asterisk-examples - PABX examples used in "Fixed and Mobile Telephone Systems" VET classes.
peerjs-server - Server for PeerJS
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