Search Results for "rtp/rtcp"
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19 programs for "rtp/rtcp" with 2 filters applied:
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The #1 Embedded Analytics Solution for SaaS Teams.Qrvey’s comprehensive embedded analytics software enables you to design more customizable analytics experiences for your end users.
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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... -
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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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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... -
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jlibrtp aims to create a library that makes it easy to support RTP (RFC 3550,3551) in Java applications. SRTP (RFC 3771) has been delayed in favor of RFC 4585.
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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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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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Java Media Library for wiring together Sinks and Sources. Can have rtp Sinks and Sources, FileRecorders/FilePlayers, Microphone, etc. The framework has default sink/sources already. It does not provide Text-to-Speech Source or Speech Recognizer Sink.
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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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Secure remote access solution to your private network, in the cloud or on-prem.OpenVPN is here to bring simple, flexible, and cost-effective secure remote access to companies of all sizes, regardless of where their resources are located.
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libfindrtp is a C library providing functions used to identify Real-time Transport Protocol (RTP) session endpoint ports and network addresses from active VoIP signaling traffic.
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The KOM(S) Streaming System (komssys) implements a streaming system based on the IETF protocols RTSP, SDP, RTP/RTCP with the intention of providing a base for researchers and other developers. Komssys includes code for a server, a client, and a proxy
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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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Speak Freely for Unix is an Internet telephony application which provides high quality voice grade audio with GSM and CELP compression and encryption with DES, Blowfish, and IDEA ciphers. It interoperates with Speak Freely for Windows and any RTP client
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This is an extension to the project OpenGatekeeper. It enable the routing through the gatekeeper of the RTP (audio and video) and T.120 (data) traffic so no traffic is directly exchanged between communication endpoints.
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This project is to extend the work of OpenH323 group to let audio/video source transmit freely between popular players in the market, f.g. media transmission between Netmeeting and Quicktime. A module to construct RTP/RTCP packet is necessary.
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