Search Results for "rtp/rtcp"

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...

REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .

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.

O'Packet Platform is a platform which aims to offer a tool that users could operate network packet (Ethernet, IP, TCP, UDP, SCTP, RTP... ) and application packet like audio/vieo packet and DTMF packet ( 2833), which refered to all rfc protocol.

Zanzibar is a complete, standards based IVR. 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.

A metering and charge-back mechanism based on real time pricing (RTP) for the cloud services. This project caters to the need of a dynamic pay-per-use model in the cloud environment.

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

RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive

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.

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.

The RTP Monitor discovers RTP flows, used by Internet phone applications, and monitors their QoS metrics, observing the interference caused by traffic congestion. It uses distributed collecting agents and provides feedback about the quality in the calls.

Software system consisting of two proxies that capture VNC over TCP traffic and converts it to VNC over RTP in order to provide streaming features to VNC.

MVIP is the multicast VRML interchange protocol. This protocol allows users to participate in shared virtual worlds. The protocol uses the services of RTP and accesses VRML through EAI.

Peer-to-Peer Videostreaming System. A normal p2p client can only download the file by using http or ftp. This System uses streaming (RTP) so that you can watch the video during the download.

Giraffe is a GUI configurator and execution tool for MARC RTP. It allows users to design and execute Request files for use with MARC RTP in an intuitive way.

The open source okeo project provides a communication platform which offer several services like Voip. The plaform can be used as a technical platform for external services using RTP/SIP and webservices.