Topic: sip Goto Github

Some thing interesting about sip

👇 Here are 587 public repositories matching this topic...

64characters / telephone

sip,SIP softphone for Mac

Organization: 64characters

Home Page: https://www.64characters.com/telephone

sip voip softphone mac-app phone-calls

8xff / atm0s-media-server

sip,Decentralized, Global-Scale Media Server written in Rust (WebRTC/Whip/Whep/Rtmp/Sip)

Organization: 8xff

Home Page: https://8xff.github.io/media-docs/

decentralized latency media-server rtmp rust scalable sfu sip streaming video

anyrtc-usecase / siprtcproxy

sip,网关服务：Sip与Rtc互通，实现Web，Android，iOS，小程序，SIP座机，PSTN电话，手机互通。

Organization: anyrtc-usecase

agora aliyun anyrtc arcall callcenter freeswitch gateway ivr kamailio pstn sip sip2webrtc tecent voip wangyiyun webrtc webrtc2sip

asterisk / asterisk

sip,The official Asterisk Project repository.

Organization: asterisk

Home Page: https://www.asterisk.org

asterisk c pbx sip voip

baresip / baresip

sip,Baresip is a modular SIP User-Agent with audio and video support

Organization: baresip

libre sip telephony voip webrtc

belledonnecommunications / linphone-android

sip,Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)

Organization: belledonnecommunications

Home Page: https://linphone.org

linphone linphone-android voip android video sip call conference phone liblinphone

belledonnecommunications / linphone-desktop

sip,Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of git://git.linphone.org/linphone-desktop.git

Organization: belledonnecommunications

Home Page: https://linphone.org/

linphone desktop voip phone sip video call conference chat message

belledonnecommunications / linphone-iphone

sip,Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of linphone-iphone (git://git.linphone.org/linphone-iphone.git)

Organization: belledonnecommunications

Home Page: http://linphone.org

linphone iphone voip phone sip video call conference chat message

caoliang1918 / contact-center

sip,智能电话外呼系统呼叫中心系统 freeswitch webrtc

User: caoliang1918

freeswitch sip pstn webrtc esl

creytiv / re

sip,Generic library for real-time communications with async IO support

Organization: creytiv

Home Page: http://creytiv.com/re.html

sip voip turn ice webrtc

datso / react-native-pjsip

sip,A PJSIP module for React Native.

User: datso

Home Page: http://datso.github.io/react-native-pjsip

react-native pjsua pjsip sip voip

drachtio / drachtio-server

sip,A SIP call processing server that can be controlled via nodejs applications

Organization: drachtio

Home Page: https://drachtio.org

sip drachtio drachtio-srf nodejs

sip,SIPVicious OSS is a VoIP security testing toolset. It helps security teams, QA and developers test SIP-based VoIP systems and applications. This toolset is useful in simulating VoIP hacking attacks against PBX systems especially through identification, scanning, extension enumeration and password cracking.

Organization: enablesecurity

Home Page: https://www.enablesecurity.com/sipvicious/oss/

sip voip password-cracker svwar svcrack svmap war-dial svcrash security security-tools audit-sip hacking-tools

flutter-webrtc / dart-sip-ua

sip,A dart-lang version of the SIP UA stack.

Organization: flutter-webrtc

voip dartlang flutter sip client jssip

flutter-webrtc / flutter-webrtc

sip,WebRTC plugin for Flutter Mobile/Desktop/Web

Organization: flutter-webrtc

android desktop flutter ios sip voip web webrtc webrtc-plugin

fonoster / routr

sip,⚡ The future of programmable SIP servers.

Organization: fonoster

Home Page: https://routr.io

sip webrtc voip instant-messaging ims asterisk kamailio freepbx freeswitch fusionpbx

gb28181 / gb28181.solution

sip,Linux/Win/Docker/kubernetes/Chart/Kustomize/GB28181/SIP/RTP/SDP/WebRTC/作为上下级域/平台级联互联

Organization: gb28181

gb28181 sip rpc sdp rtsp csharp linux dotnetcore dotnet-core gb28181-2016

ghettovoice / gosip

sip,SIP in Go

User: ghettovoice

golang hacktoberfest sip

havfo / webrtc-to-sip

sip,Setup for a WEBRTC client and Kamailio server to call SIP clients

User: havfo

webrtc-client rtpengine kamailio bridge sip webrtc turn

hnimminh / libresbc

sip,An open source Session Border Controller 🌟 The SBC you dream about 🗽 LibreSBC will help you save thousands of dollars.

User: hnimminh

Home Page: https://github.com/hnimminh/libresbc

sbc freeswitch asterisk kamailio freepbx fusionpbx opensips sip sip-server session-border-controller

infactum / tg2sip

sip,Telegram <-> SIP voice gateway

User: infactum

sip telegram asterisk freeswitch pjsip

innovateasterisk / browser-phone

sip,A fully featured browser based WebRTC SIP phone for Asterisk

User: innovateasterisk

Home Page: https://www.innovateasterisk.com

webrtc browser-phone asterisk-pbx audio-calls video-calls voip asterisk asterisk-server asterisk-dialplan asterisk-webui

ireader / media-server

sip,RTSP/RTP/RTMP/FLV/HLS/MPEG-TS/MPEG-PS/MPEG-DASH/MP4/fMP4/MKV/WebM

User: ireader

flv fmp4 hls m3u m3u8 mkv mp4 mpeg-dash ps rtmp rtp rtsp sdp sip ts webm

irontec / sngrep

sip,Ncurses SIP Messages flow viewer

Organization: irontec

sip debug trace flow terminal pcap hep ncurses

jfaler / soup

sip,☎️ Original open source call flooder using Twilio's API.

User: jfaler

Home Page: https://justinfaler.com

twilio call api pbx sip trunk phone tdos ddos hacking

jitsi / jitsi

sip,Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.

Organization: jitsi

Home Page: https://desktop.jitsi.org

jitsi icq jabber java xmpp instant-messaging xmpp-client audio video irc

kamailio / kamailio

sip,Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -

Organization: kamailio

Home Page: https://www.kamailio.org

kamailio sip voip webrtc volte iot telephony

meliht / mr.sip

sip,SIP-Based Audit and Attack Tool

User: meliht

sip voip audit pentest dos ip-spoofing attack ddos

miconda / sipexer

sip,Modern and flexible SIP/VoIP cli tool

User: miconda

go golang sip telephony voip webrtc websocket

mikopbx / core

sip,Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core

Organization: mikopbx

Home Page: https://www.mikopbx.com

asterisk pbx telephony phalcon cti pjsip voip-server voip asterisk-pbx asterisk-ami

miroslavpejic85 / mirotalksfu

sip,🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.

User: miroslavpejic85

Home Page: https://sfu.mirotalk.com

sfu webrtc self-hosted voip chat docker video audio mediasoup scalable-video-conferences

onmyway133 / awesome-voip

sip,🤙Learning VoIP, RTP, pjsip and SIP

User: onmyway133

Home Page: https://onmyway133.github.io/

c sip voip

onsip / sip.js

sip,A simple, intuitive, and powerful JavaScript signaling library

User: onsip

Home Page: https://sipjs.com

sipjs webrtc javascript sip voip typescript nodejs

panjjo / gosip

sip,sipserver,GB28181,ZLMediaKit

User: panjjo

sip gb28181 zlmediakit sipserver

paullouisageneau / libjuice

sip,JUICE is a UDP Interactive Connectivity Establishment library

User: paullouisageneau

ice udp sdp stun nat-traversal nat-hole-punching p2p peer-to-peer sip webrtc

pepelux / sippts

sip,Set of tools to audit SIP based VoIP Systems

User: pepelux

voip hacking pentesting secutiry asterisk freepbx elastix pentest pentest-tool hacking-tool

pion / stun

sip,A Go implementation of STUN

Organization: pion

Home Page: https://pion.ly/

go golang stun voip ice sip webrtc networking peer-to-peer streaming

pjsip / pjproject

sip,PJSIP project

Organization: pjsip

Home Page: http://www.pjsip.org

pjsip sip rtp nat-traversal voip android ios android-ndk

processone / ejabberd

sip,Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)

Organization: processone

Home Page: https://www.process-one.net/en/ejabberd/

chat ejabberd erlang groupchat iot jabber matrix messaging mqtt pubsub server sip stun turn voip xmpp

restcomm / restcomm-connect

sip,The Open Source Cloud Communications Platform

Organization: restcomm

Home Page: http://www.restcomm.com/

restcomm java communication voip sip

rtckit / awesome-rtc

sip,:satellite: A curated list of awesome Real Time Communications resources

Organization: rtckit

awesome rtc awesome-list sip voip telephony webrtc telecommunications real-time-communications

saghul / sipp-scenarios

sip,SIPp scenarios I use for testing SIP stuff

User: saghul

Home Page: http://bettercallsaghul.com

sip sipp

signalwire / freeswitch

sip,FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.

Organization: signalwire

Home Page: https://freeswitch.com/#getting-started

freeswitch iot open-source signalwire sip source-code telephony voip webrtc