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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...
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/
Simple testing tool to generate RTP data packets and send it via netwok interface or save into pcap file. Primarily intended for use with SIPp application to test speech quality with different codecs.
Steganography tool which establishes a full-duplex steganographic data transfer protocol utilizing Real-time Transfer Protocol (RTP) packet payloads as the cover medium. The tool provides interactive chat, file transfer, and remote shell access.
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Shared white board using SVG and RTP is an attempt to standardize transport medium, data format and communication protocol for shared drawing sessions. For more details see: http://www1.cs.columbia.edu/~gg2265/report/SharedWhiteBoard_SVG_RTP.htm Browse c
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.
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
Enable Linux firewall to support connection tracking and NAT of H.323 protocol. It supports RAS, Fast Start, H.245 Tunnelling, Call Forwarding, Signal Proxy/Softswitch, RTP/RTCP and T.120 based audio, video, fax, chat, whiteboard, file transfer, etc.
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.
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
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.