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Show HN: Vialer-js – Open-source WebRTC communication platform

127 点作者 jvanveen将近 7 年前

7 条评论

fenesiistvan将近 7 年前
I am a VoIP developer and my recommendation for WebRTC is to just use any legacy Softswitch or IP-PBX. These are more matured software, with tons of features and all of them has support (also) for WebRTC. Most of them has much better performance than Node.js implementations (you can handle more simultaneous clients). Both open-source and commercial software is available with good support:<p>-Asterisk: <a href="https:&#x2F;&#x2F;wiki.asterisk.org&#x2F;wiki&#x2F;display&#x2F;AST&#x2F;Asterisk+WebRTC+Support" rel="nofollow">https:&#x2F;&#x2F;wiki.asterisk.org&#x2F;wiki&#x2F;display&#x2F;AST&#x2F;Asterisk+WebRTC+S...</a><p>-FreeSwitch: <a href="https:&#x2F;&#x2F;freeswitch.org&#x2F;confluence&#x2F;display&#x2F;FREESWITCH&#x2F;WebRTC" rel="nofollow">https:&#x2F;&#x2F;freeswitch.org&#x2F;confluence&#x2F;display&#x2F;FREESWITCH&#x2F;WebRTC</a><p>-Mizutech: <a href="https:&#x2F;&#x2F;www.mizu-voip.com&#x2F;Software&#x2F;VoIPServer.aspx" rel="nofollow">https:&#x2F;&#x2F;www.mizu-voip.com&#x2F;Software&#x2F;VoIPServer.aspx</a><p>If you choose a softswitch which doesn’t have built-in WebRTC support, you can easily add it using a WebRTC-SIP gateway module:<p>-Doubango: <a href="https:&#x2F;&#x2F;github.com&#x2F;DoubangoTelecom&#x2F;webrtc2sip" rel="nofollow">https:&#x2F;&#x2F;github.com&#x2F;DoubangoTelecom&#x2F;webrtc2sip</a><p>-MRTC: <a href="https:&#x2F;&#x2F;www.mizu-voip.com&#x2F;Software&#x2F;WebRTCtoSIP.aspx" rel="nofollow">https:&#x2F;&#x2F;www.mizu-voip.com&#x2F;Software&#x2F;WebRTCtoSIP.aspx</a><p>-Janus: <a href="https:&#x2F;&#x2F;janus.conf.meetecho.com&#x2F;" rel="nofollow">https:&#x2F;&#x2F;janus.conf.meetecho.com&#x2F;</a><p>On the client side you can use any RFC 7118 compilant WebRTC client:<p>-SIPML5: <a href="https:&#x2F;&#x2F;www.doubango.org&#x2F;sipml5&#x2F;" rel="nofollow">https:&#x2F;&#x2F;www.doubango.org&#x2F;sipml5&#x2F;</a><p>-WebPhone: <a href="https:&#x2F;&#x2F;www.mizu-voip.com&#x2F;Software&#x2F;WebPhone.aspx" rel="nofollow">https:&#x2F;&#x2F;www.mizu-voip.com&#x2F;Software&#x2F;WebPhone.aspx</a><p>-SIP.js: <a href="https:&#x2F;&#x2F;sipjs.com&#x2F;" rel="nofollow">https:&#x2F;&#x2F;sipjs.com&#x2F;</a>
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jvanveen将近 7 年前
Author here: The first priority of the project is to offer a solid generic audio-only calling experience to our users. This includes all functionality that people would expect from a softphone. At the same time, we want to build something that is generic and useful to any SIP-over-wss provider in the short term. The build mechanism and module-system should accomodate this. It is still under construction (<a href="https:&#x2F;&#x2F;github.com&#x2F;vialer&#x2F;vialer-js&#x2F;tree&#x2F;feature&#x2F;restructure-module-loading" rel="nofollow">https:&#x2F;&#x2F;github.com&#x2F;vialer&#x2F;vialer-js&#x2F;tree&#x2F;feature&#x2F;restructure...</a>), but most of the hardcoded dependencies are already dealt with.<p>Besides the current SIP Call implementation, we are preparing the codebase for additional signalling implementations as well. The centralized SIP Call implementation is great to use with all the functionalities a PBX offers (queues, on-hold, transfers, being available to any device), but browser-to-browser p2p also has its strong points. Privacy-aware chat, video and file transfer seem easier to implement and maintain with nothing more than JavaScript and another browser as a backend. The signalling part will be a bit of a design challenge. Ideally, this would be p2p and encrypted too, so people don&#x27;t need to rely on one server&#x2F;provider to find their contacts and setup connections with.
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sam_goody将近 7 年前
There are a few open source RTC platforms.<p>On first blush, Vialer seems to be softphone centric - audio only, mostly peer to peer, softphone features. These others target one to many video streaming.<p>Is that right?<p>- <a href="https:&#x2F;&#x2F;www.kurento.org" rel="nofollow">https:&#x2F;&#x2F;www.kurento.org</a><p>- <a href="https:&#x2F;&#x2F;mediasoup.org" rel="nofollow">https:&#x2F;&#x2F;mediasoup.org</a> (for node)<p>- <a href="https:&#x2F;&#x2F;janus.conf.meetecho.com" rel="nofollow">https:&#x2F;&#x2F;janus.conf.meetecho.com</a><p>- <a href="https:&#x2F;&#x2F;github.com&#x2F;opentok&#x2F;OpenTokRTC-V2" rel="nofollow">https:&#x2F;&#x2F;github.com&#x2F;opentok&#x2F;OpenTokRTC-V2</a><p>OT: is there a way to do markdown in HN comments?
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kodablah将近 7 年前
Does signalling phone home? Would be neat to do serverless signalling. It might be doable by putting the offer in the URL (and letting people text it to each other) or maybe packing the offer in a QR code or maybe using a DHT inside of JS. I have been toying w&#x2F; the IPFS JS impl, and have successfully connected to the DHT and done lookups over web sockets, but I would assume that the webrtc transport would allow you to participate in the DHT.
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benoits将近 7 年前
If any senior VoIP&#x2F;SIP&#x2F;WebRTC dev is interested, my telemedicine startup just won the tender for building the WebRTC platform used by all hospitals in the Paris region. Includes some cool medical devices data streaming too. We&#x27;re desperately understaffed so we&#x27;re offering very competitive compensation, PM me. Stack is React+Node.js+Kurento but could change based on your input.
Fifer82将近 7 年前
Does anyone have time to ELI5 WebRTC?
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pierut将近 7 年前
hey why does the web need a real-time clock