Search Results for "rtp sip"
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Showing 18 open source projects for "rtp sip"
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. ... -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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Xplico
Xplico is a Network Forensic Analysis Tool (NFAT)
...The goal of Xplico is extract from an internet traffic capture the applications data contained. For example, from a pcap file Xplico extracts each email (POP, IMAP, and SMTP protocols), all HTTP contents, each VoIP call (SIP, MGCP, MEGACO, RTP), IRC, WhatsApp... Xplico is able to classify more than 140 (application) protocols. Xplico cam be used as sniffer-decoder if used in "live mode" or in conjunction with netsniff-ng. Xplico is used also in CapAnalysis: http://www.capanalysis.net -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call Centers and Contact Centers for Quality monitoring (QM) purposes. -
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TOFFEE-DataCenter
Linux Server and Data-Center WAN Optimization
TOFFEE-DataCenter is the new TOFFEE variant meant for Data-Center, Server, HPC, Load Balancing, Cluster/Cloud Computing and SDN deployments. With TOFFEE-DataCenter (and or TOFFEE), OEMs and other equipment vendors can make use of this platform/stack, integrate, port and build WAN Optimization devices/appliances on to their existing product range. Or build a dedicated WAN Optimization hardware product series via the same. -
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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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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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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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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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...It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.
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SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi
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This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.
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A plugin for Windows ALG Proxy Service to modify SIP/RTP packets. Currently it doesn't work and needs other developers to help.
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A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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An IP Based decentralised Call recorder in Java Main Components: Packet Sniffer (included offline tcpdump file andalyzer) , SIP/RTP Analyzer (separates SIP and RTP packets from the stream , Network Mapper (Draws a diagram of the nodes)
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simpleRTSP is a compact,fast,lightweight RTSP libraries,using open standard protocols(RTP/RTCP,RTSP,SIP).It aims to provide simple and easy-ported C libraries for embedded environment,such as set-top box,hand-held terminals and digital-media devices.
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Simple command line IP telephonly application for Linux, that doesn't conform to either H.323 or SIP, doesn't use RTP and is not interoperable with any other program. It does work well though, and has quite good audio quality.
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