Pipes a WebRTC video stream to a video element. el can be a string (id), jquery element, or dom node.
http://github.com/wearefractal/hollaTags | webrtc rtc calling call voip p2p data sip voice phone peer |
Implementation | Javascript |
License | MIT |
Platform | NodeJS |
Linphone is an internet phone or Voice Over IP phone (VoIP), it helps to communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone is available for Linux, Windows, MacOSX, and for mobile phones: Android, iPhone, Blackberry. It is using SIP protocol and it is compatible with any voip operator using SIP for its network.
voip skype-alternative internet-phone sip xmpp chatSFLphone is a softphone for Linux. It is SIP compatible and supports Multi-way conferencing, Peer to peer calls, Video calls, Instant Messaging, Multi-line, Call transfer, Call hold, Call recording and lot more. It encrypts voice using SRTP protocol.
voip skype-alternative internet-phone chat sip softphoneWebRTC SDK Upgraded! ES6, new camera control and 100x less code than v1. At PubNub we believe simplicity is essential for our SDK usability. We've taken a simplified approach to WebRTC Peer Connections by creating and easy-to-use SDK for developers. The ideas of simplicity should span all platforms and devices too and that's why we also support Android WebRTC mobile calling with compatibility for iOS native Objective-C based WebRTC SDK. This simple developer WebRTC SDK is powered by PubNub Data Stream Network.
webrtc webrtc-call webrtc-sdk webrtc-video webrtc-receiver webrtc-javascript-library webrtc-phone webrtc-demo webrtc-dialingThis module works in the browser with browserify.Note: If you're NOT using browserify, then use the included standalone file simplepeer.min.js. This exports a SimplePeer constructor on window.
webrtc p2p nodejs browser data-channels peer-connection data data-channel data-channel-stream peer peer-to-peer stream video voice webrtc-streamAn abstraction over P2P video/voice/data connections using WebRTC
Kamailio (successor of former OpenSER and SER) is an open source implementation of a SIP Signaling Server. SIP is an open standard protocol specified by the IETF. The core specification document is RFC3261.
kamailio sip voip webrtc volte iot telephony sip-serverHOMER is a robust, carrier-grade, scalable SIP Capture system and VoiP Monitoring Application offering HEP/EEP, IP Proto4 (IPIP) encapsulation & port mirroring/monitoring support right out of the box, ready to process & store insane amounts of signaling, logs and statistics with instant search, end-to-end analysis and drill-down capabilities for ITSPs, VoIP Providers and Trunk Suppliers using SIP signaling protocol. Powered at the core by SIPCAPTURE Module for industry-standard Kamailio or OpenSIPS, HOMER provides virtually unlimited scope for granular capture configuration either stand-alone or using our companion Capture Agent Project.
monitoring troubleshooting voip rtc pcap flow callflow cdr correlation capture-agent packet-capture packet-sniffer analytics webrtc encapsulation sip statistics opensips kamailio hepTelephone is a VoIP program which allows you to make phone calls over the internet. It can be used to call regular phones via any appropriate SIP provider. If your office or home phone works via SIP, you can use that phone number on your Mac anywhere you have decent internet connection. Opus codec is optional.
sip voip softphone mac-app phone-callsAsterisk, converts an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services, including IP PBXs, VoIP gateways, call center ACDs and IVR systems. It is maintained by Debian VoIP Team.
communication framework ip-telephony pbx voip call-centerQuteCom is a SIP compliant VOIP client. It allows users to speak to other users of SIP compliant VoIP software. It helps to make calls to landlines, cellphones and allows to send SMS and make video calls.
voip skype-alternative internet-phone chat sip softphoneAll domain names that are not bound in the management system (https://oms.cdnbye.com) will stop providing P2P services. Please bind your domain names in time to avoid being affected. Put the quick-start.html in your web page, run it. Wait for a few seconds,then open the same page from another browser. Now you have a direct P2P connection between two browsers without plugin! The first web peer will serve as a seed, if no one else in the same channel.
html5 webrtc video mse player p2p hls live vod cdnbye cdn peer peer-to-peer bittorrent webrtc-data-channel http-live-streamingBasically, it is a telecommunication module which handles most of requirements when making/receiving/talking with a call. note: you might need android.permission.BLUETOOTH permisions for Bluetooth to work.
react reactnative webrtc phone call incall talk sip voipA proof-of-concept for WebRTC signaling using sound. Works with all devices that have microphone + speakers. Runs in the browser. Nearby devices negotiate the WebRTC connection by exchanging the necessary Session Description Protocol (SDP) data via a sequence of audio tones. Upon successful negotiation, a local WebRTC connection is established between the browsers allowing data to be exchanged via LAN.
data-over-sound webrtc fsk p2p webrtc-signaling file-sharing data-transfer modem ultrasonicEkiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet. It provides Audio and Video free calls through the internet. It supports standard telephony features like Call Hold, Call Transfer, Call Forwarding, Call Histroy and Call Monitoring.
voip skype-alternative internet-phone sip chatLinphone is a free VoIP and video softphone based on the SIP protocol. Download the Android sdk (API 28.0.0 at max) with platform-tools and tools updated to latest revision, then add both 'tools' and 'platform-tools' folders in your path and the android-sdk folder to ANDROID_HOME environment variable.
linphone linphone-android voip android video sip call conference phone liblinphoneRoll Call is a completely free🎉 voice chat service with podcast quality recording. For more information on how to use Roll Call check out the FAQ.
webrtc p2p nodejs privacyThis module provides an easy and reliable way to setup a WebRTC connection between peers and communicate using events (the socket.io-protocol). Socket.IO is used to transport signalling data and as a fallback for clients where WebRTC PeerConnection is not supported.
webrtc peer-to-peerBlink is a GUI for Mac, Windows and Linux built on top of SIP SIMPLE client SDK. It uses SIP (Session Initiated Protocol) for VOIP and Instant Messaging. It supports Multi-party conferencing, Missed Calls, Dialing out SIP addresses and telephone numbers. Currently File Transfer and Desktop sharing is supported only in Mac.
voip skype-alternative internet-phone sip chat softphoneCacheP2P is a highly distributed cache platform based on WebTorrent and runs only in the browser. It is a javascript library that once included in a website, makes every new user a mirror of the specific URL he has opened and allows it to serve it to all the other users that also are accessing the same website, so the website's server doesn't have to.
cache p2p distributed webtorrent web2web cachep2p p2pcache web2webcache cacheweb2web bittorrent bittorrent-client download mad-science peer-to-peer peers streaming swarm torrent web-torrent webrtc webrtc-data
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