Search Results for "rtp/rtcp"
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Showing 11 open source projects for "rtp/rtcp"
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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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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... -
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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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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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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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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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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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