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Search results for sip rtp

76 search results found

Media Server ⭐ 2,818

RTSP/RTP/RTMP/FLV/HLS/MPEG-TS/MPEG-PS/MPEG-DASH/MP

Pjproject ⭐ 1,751

Baresip ⭐ 1,559

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

Sipsorcery ⭐ 1,299

A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.

Gb28181.solution ⭐ 523

Linux/Win/Docker/kubernetes/Chart/Kustomize/GB2818

Generic library for real-time communications with async IO support

Pentest101 ⭐ 456

一些关于渗透测试的Tips

Public Switched Telecommunications Network Unleashed

Rtpproxy ⭐ 385

The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.

Gb28181 Server ⭐ 260

LiveGBS国标(GB28181)流媒体服务软件：提供用户管理及Web可视化页面管理；提供设备状态管理，可实时查看设备是否掉线等信息；实时流媒体处理，PS（TS）转ES；设备状态监测、云台控制、录像检索、回放；提供RTSP、RTMP、HTTP-FLV、HLS等多种协议流输出；对外提供服务器获取状态、信息，控制等HTTP API接口；支持语音对讲；支持云端录像；TCP、UDP两种方式信令传输以及UDP、TCP被动、TCP

Webrtc To Sip ⭐ 255

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

Jazminserver ⭐ 247

Java based application,rpc,message,rtmp,game,sip,rtp,relay,we server,message queue,mysql proxy server

Sniffer ⭐ 210

VoIPmonitor sniffer sources

Homer App ⭐ 186

HOMER 7.x Front-End and API Server

Pure python VoIP/SIP/RTP library. Currently supports PCMA, PCMU, and telephone-event

Java SIP softphone

Awesome Rtc ⭐ 161

📡 A curated list of awesome Real Time Communications resources

Wirebug ⭐ 142

WireBug is a toolset for Voice-over-IP penetration testing

Raspberry Pi RTP MIDI

Skyway Webrtc Gateway ⭐ 130

WebRTC Gateway for SkyWay

Generic library for real-time communications with async IO support

Xswitch Free ⭐ 109

Free XSWITCH Docker Image and More ... | 免费的XSWITCH镜像及其它

Sip3 Ansible ⭐ 105

Ansible scripts to install and configure SIP3

Voip_patrol ⭐ 95

VoIP signaling and media test automation

Pyfreebilling ⭐ 87

P-KISS-SBC - simple and stupid SIP/RTP SBC - AGPL v3 - Based on kamailio / RTP Engine

Homer7 Docker ⭐ 87

HOMER 7 Docker Images

Barebone Opensource Powered SBC

Sip3 Salto Ce ⭐ 76

SIP3 Salto (Community Edition)

Sip3 Captain Ce ⭐ 74

SIP3 Captain (Community Edition)

Light weight implementations of real-time communication protocols and applications in Python

Sip3 Twig Ce ⭐ 61

SIP3 Twig (Community Edition)

Project of Multi-protocol Test Tool opensourced by Ericsson

Asterisk Config ⭐ 46

Kubernetes dynamic configuration engine for Asterisk

Sipp By Example ⭐ 44

Mediaproxy ⭐ 41

Media relay for RTP/RTCP and UDP streams

Awesome Voip ⭐ 40

🤙Learning VoIP, RTP, pjsip and SIP

Cppsipstack ⭐ 38

FreePBX container (Asterisk 16; OpenPBX 15 with Backup and IVR modules installed)

Python Sipsimple ⭐ 33

Mirror repositorie from darcs. SIP SIMPLE client SDK.

Vag.node ⭐ 31

GB28181 PS流转发网关服务<Node 版>，以GB28181对接的方式将摄像机/硬盘录像机的PS流（H264/H265）打包推送到RTMP服务器。

Kurento Sip Gw ⭐ 29

Drachtio Rtpengine Webrtcproxy ⭐ 29

Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)

Linphone ⭐ 29

mirror of git clone git://git.linphone.org/linphone.git --recursive

Freeswitch Docker ⭐ 29

Dockerfile for freeswitch

Viva Voce ⭐ 28

P2P VoIP app in C++ using ADPCM-encoded audio, SIP, SDP and RTP over UDP/IP.

Sip3 Documentation ⭐ 28

SIP3 Documentation

VoIPong is a utility which detects all Voice Over IP calls on a pipeline, and for those which are G711 encoded, dumps actual conversation to seperate wave files. It supports SIP, H323, Cisco's Skinny Client Protocol, RTP and RTCP.

Sipstack ⭐ 25

A SIP stack for Delphi, including SDP parsing and an RTP stack.

Rtpproxy ⭐ 25

RTPProxy - application for RTP packets relaying - additional patches

Toolkit library for asynchronous network IO with protocol stacks including SIP, SDP, RTP, STUN, TURN, ICE, BFCP and DNS.

Wsbridge ⭐ 22

Platform-independent voice-over-IP peer-to-peer communication program

Sip Voice Quality Report Reaper ⭐ 17

The SIP Voice Quality Report Reaper sniffs RTCP and RTP packets and generates SIP PUBLISH messages with voice quality reports.

Non-official repository of old released libre packages

Skype Kamailio Pstn Gateway ⭐ 16

Skype for Business/Lync PSTN gateway

Kamailio Static Relay ⭐ 16

A simple SIP and RTP relay configuration with static routing

Sip2rtsp ⭐ 14

sip/ims protocol to rtsp protocol gateway

Awesome Hep ⭐ 13

A curated list of HEP / EEP enabled projects

ohrwurm is an RTP fuzzer. features some SIP parsing and RTCP suppressing.

Testprogram for libre and librem

A straight-forward SIP/SDP parsing module for Node/JS

Telegram <-> SIP voice gateway

Sip2ban.github.io ⭐ 9

Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS

Lib Mm Live555_streaming_media ⭐ 7

This code forms a set of C++ libraries for multimedia streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These libraries - which can be compiled for Unix (including Linux and Mac OS X), Windows, and QNX (and other POSIX-compliant systems) - can be used to build streaming applications.

The SIP Voice Quality Report Reaper sniffs RTCP and RTP packets and generates SIP PUBLISH messages with voice quality reports.

Docker Rtpengine Speech ⭐ 7

OpenSIPS + RTPEngine Recording + Speech Recognition in HEP

IP SLA is a project to control the SLA of VoIP circuits

High-performance Session Initiation Protocol (RFC3261) daemon.

Jabber4linux ⭐ 6

Unofficial Cisco Jabber Softphone Implementation for Linux

Akstream Readme ⭐ 6

Go Baresip ⭐ 5

Go baresip wrapper for automated SIP tests

Wireshark Opus ⭐ 5

Parses a Wireshark PDML file that contains an RTP+Opus SIP call and outputs wave audio files

Kms Siprtpendpoint ⭐ 5

Kurento SipRtpEndpoint modiule, provides the capabilities of connecting RTP media flows from a SIP network into Kurento. Of course it will require some SIP signalling plane not included in this module.

Microsippy ⭐ 5

Extremely fast and lean, yet fully functional, SIP (RFC3261) and RTP (RFC3550) implementation

Webrtc Gateway to SIP/RTP (experimental work in progress)

Ecg_extract_call ⭐ 5

Extract VoIP (SIP) Call Signaling and Corresponding RTP

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