Search Results for "rtp/rtcp"
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Showing 16 open source projects for "rtp/rtcp"
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From improving customer experience through seamless sign-on to making MFA as easy as a click of a button – your login box must find the right balance between user convenience, privacy and security.
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ConnectWise CPQ, formerly ConnectWise Sell, is a professional quote and proposal automation software for IT solution providers. ConnectWise CPQ offers a wide range of tools that enables IT solution providers to save time, quote more, and win big. Top features include professional quote or proposal templates, product catalog and sourcing, workflow automation, sales reporting, and integrations with best-in-breed solutions like Cisco, Dell, HP, and Salesforce.
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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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SipTunnel is used to tunnel SIP UDP datagrams (and also RTP datagrams) through NAT using TCP protocol. It also can be used for many other purposes.
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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... -
ConnectWise PSA (formerly Manage) allows you to stop working in separate systems, and helps you build a more profitable business. No more duplicate data entries, inefficient employees, manual invoices, and the inability to accurately track client service issues. Get a behind the scenes look into the award-winning PSA that automates processes for each area of business: sales, help desk, support, finance, and HR.
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
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. . -
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This application use sipp. The GUI base application try to create xml and csv files easily and start scenario which are selected. sipp and mono have to be installed on your PC. If you want to send RTP packets, you should copy pcap files to same folder where running sipp_gui.
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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.
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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.
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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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When it comes to cybersecurity, what your clients don’t know can really hurt them. And believe it or not, keep them safe starts with asking questions. With ConnectWise Identify Assessment, get access to risk assessment backed by the NIST Cybersecurity Framework to uncover risks across your client’s entire business, not just their networks. With a clearly defined, easy-to-read risk report in hand, you can start having meaningful security conversations that can get you on the path of keeping your clients protected from every angle. Choose from two assessment levels to cover every client’s need, from the Essentials to cover the basics to our Comprehensive Assessment to dive deeper to uncover additional risks. Our intuitive heat map shows you your client’s overall risk level and priority to address risks based on probability and financial impact. Each report includes remediation recommendations to help you create a revenue-generating action plan.
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The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.
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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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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
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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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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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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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