Search Results for "sip" - Page 3
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Showing 221 open source projects for "sip"
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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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Steve Murray Phone
simple SIP video phone source/lib/dll/API with h264, ulaw and g729
I coded up this simple SIP engine and video phone a few years ago. I realized how easy it is to speak SIP without a lot of classes or abstraction. It is just plain-english SIP. The SIP is copy-pasted from wireshark, and then just does replace of whatever characters need to be changed to make or answer a specific call. I don't program much VOIP any more, and pretty soon I might have thrown away all my hard drives with this code on it. -
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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SIPFlow
Capture and display SIP and DIAMETER messages
...This allows network engineers to quickly identify the behavior of their IMS network without tracing through log files or raw captures. Distributed SIPFlow captures and displays SIP and Diameter messages from remote hosts in an intuitive graphical interface. Each SIPFlow console interfaces with one or more SIPSniffers distributed across a SIP/IMS network and displays the SIP and Diameter call flows as reported by those sniffers. This is useful if your network of servers span multiple network segments or switches. ... -
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JSIP2
SIP2 Library for Java.
3M Standard Interchange Protocol (SIP) is an industry standard protocol by 3M to allow automatec check out terminals communicate with library systems. JSIP2 is an Open Source (MIT) library that implements the 3M SIP version 2 protocol including the 3M SIP2 Extensions for SIP2 clients. The library contains the necessary functionality for SIP2 client socket based implementation supporting all the SIP2 messages and fields, including also the fields defined in the the 3M SIP2 Extensions. -
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).
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Sippy B2BUA and RFC3261 SIP Stack
This project has been relocated to http://github.com/sippy/rtpproxy h
This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua -
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Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
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SSL Diagnos
SSL Strength Evaluation and Test Utility
SSL Diagnos is used to test SSL strength; get information about SSL protocols (pct, ssl2, ssl3, tls, dtls) and cipher suites. It can also be used for testing and rating ciphers on SSL clients. It has also specific support for pop3s, sip, smtp and explicit ftps. Tests for heartbleed (including dtls). Furthermore a separate tool, SSLPressure, not using openssl can be used to check the whole spectrum of possible SSL protocols on a server. Can also be used for testing ssl for mssql-servers (was added since nessus did not support this) and contains mitm poc for stripping ssl from mssql-connections. -
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for example, HTTP servers). ... -
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Artemisa is a honeypot for VoIP (SIP) networks. It is designed to connect to a VoIP domain as a user-agent backend in order to detect malicious activity at an early stage and also adjust the policies of the enterprise in real-time.
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OpenBFCPSipClient
OpenBFCPSipClient External Libraries and Sources
본 프로젝트는 하드웨어 화상회의 장비와 영상회의 및 디지털 협력 가능한 소프트웨어 단말이다. 1. 영상회의(음성,영상) 기능 - 콜매니저(Call Manager) : SIP, RTP 스택을 수용하고, PC영상 수신 제어를 위해 BFCP 스택을 수용한다. - 듀얼 비디오 스트리밍 모듈 : 카메라 영상 이외에 추가적인 PC영상을 수신하는 기능을 수행한다. - 미디어 코덱 : 실시간으로 음성(G.711)과 영상(H.264)을 인코딩/디코딩하는 기능을 수행한다. 2. 디지털 협업 기능 - 패킷 포맷 관리 모듈 : BFCP 패킷을 분석하고 정보를 관리하는 모듈이다. - 참여관리 모듈 : 회의 주최자와 참여자 사이의 동작을 제어하는 모듈이다. - 클라이언트/서버 모듈 : BFCP 연결 관장 및 인증을 관리하는 모듈이다. -
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rtplib
Tiny library for RTP Voice
This Tiny RTP library for alaw and ulaw voice transmission for SIP -
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ugent sip creator
graphical user interface to create bagits
This software provides a graphical user interface for these tree tasks: 1. perform renaming operation on files 2. create bagits, with a mets.xml as tag file to describe the metadata 3. create an export of this bagit for a specific digital repository that does not support bagit (e.g. DSpace). -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git
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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
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A simple interface to work with SIP numbers of Asterisk. Add, delete, and edit SIP numbers in the MySQL database As well as viewing CDR records Простой интерфейс для работы с SIP номерами сервера Asterisk. Добавление, удаление и редактирование SIP номеров в базе MySQL А так же просмотр CDR записей
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DragonEyes
An application for multiparty video/audio conference
The DragonEyes is a linux software which enables multimedia, multiparty collaboration in applications such as group conferencing, distance learning, training and video telephony. All written in C++ and with a GUI frontend. The interface is based on Qt. The media backend is based on the GStreamer framework. Communication protocols include SIP, RTP. -
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pjSIP_Recovery
vb .net wrapper for pjSIP project featuring enhanced call recovery
vb .net wrapper for pjSIP project featuring enhanced call recovery features. This project has a ,net wrapper with extended sip functionality wrappered around call recovery features such as rtp timeout and call redelivery. -
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SIP UMS is an Unified Messaging Server for the SEMS/IVR. It is an extendable voicemai and IVR platform written in Perl. It includes a web based adminstration tool, voice prompts (in english) and sample scripts that will work with SEMS/IVR.
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SIP-Validator ist a tool that sniffs SIP-messages from network and checks the syntax of the message-headers corresponding to RFC3261.
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In this project a real time VoIP signaling analyzer and alarming system was developed which deals with SIP signaling. . The popular C library WinPcap was used to build up the program. The packets were stored in MySQL database through MySQL shell.
