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Search results for webrtc asterisk

17 search results found

Routr ⭐ 1,273

⚡ The future of programmable SIP servers.

Browser Phone ⭐ 379

A fully featured browser based WebRTC SIP phone for Asterisk

Awesome Rtc Hacking ⭐ 252

a list of awesome resources related to security and hacking of VoIP, WebRTC and VoLTE

Saraphone ⭐ 96

SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. SaraPhone is fully integrated with FusionPBX. Based on SIP.js, SaraPhone works with all WebRTC compliant servers: FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc. SaraPhone gets its name from Giovanni's wife, Sara.

Definitive Guide 5th Edition ⭐ 45

Перевод замечательной книги Asterisk™: Полное руководство, 5-е издание

Webphone Sip ⭐ 31

WebRTC SIP based VoIP client software (+chrome extension)

Ominicontacto ⭐ 26

The Open Source Contact Center Solution (mirror of https://gitlab.com/omnileads/ominicontacto)

Pbxwebphone ⭐ 25

WebRTC based webphone for Vicidial

Webrtc Sip Example ⭐ 23

A small example of how to build a WebRTC application using SIP as signaling layer

Dana The Stream Gatekeeper ⭐ 21

React based front-end demo for Asterisk's SFU

Chan_respoke ⭐ 14

Asterisk Channel Driver for Respoke

Asterisk Sipml5 ⭐ 12

A complete guide to install Asterisk and use sipml5 with python server. Developed for Audio call using webrtc js library sipml5 and Asterisk's Pjsip.

Jsep To Sip Gw ⭐ 10

JSEP to SIP gateway to allow webrtc clients to talk to SIP clients

Awesome Conference ⭐ 6

An Asterisk/ARI/Respoke WebRTC video-presenter conference

Wtkrtc Sdk Api ⭐ 6

WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.

Module of FreePBX (WebRTC Phone) :: The WebRTC Module allows an Administrator to enable a "WebRTC phone" that can be attached to a user's extension which they can connect to through FreePBX User Control Panel, this WebRTC phone will then receive phone calls at the same time as the users extension using user and device mode behind the scenes. If you have User and Device Mode enabled any extension you enable the WebRTC Phone a duplicate extension of 99XXXX will be created (where XXXX is the origin

Module of FreePBX (Certificate Manager) :: Certificate Manager for Asterisk. Used for TLS, DTLS connection (think WebRTC and secure traffic)

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1-17 of 17 search results

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