Search Results for "audio server"

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...

JavaComm is a client–server communication platform offering chat, telephony, screen sharing, conferencing, a media player, and configurable media streams. Users can define and manage their own streams for flexible playback. All server-based communication runs over HTTPS (SSL/TLS), while file transfers are handled directly between users without passing through the server. The JavaComm Server runs on any Jakarta EE Web Profile 10 platform with the required extensions, using Tomcat 10.1.54 as...

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...

JAVIER is a JAvascript Voicexml InterpretER, designed (but not restricted) to run inside a web browser, its main engine has less than 1000 lines of code. It's maybe, the tiniest but (almost) FULL VoiceXML implementation.

A plugin for Teamspeak3. This plugin allows you to follow a user while he switches through channels. For the love menu just right click any name in the server view.

A collection of software made by Milos Rancic.

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.

VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K

CHAROZT is an Asterisk MOH (Music on Hold) Uploader. It is a Linux-based application installed in Linux desktops that converts MP3 to WAV for Asterisk supported MOH. It can also upload converted files to the Asterisk Server.

IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.

Bayonne is supported from Savanna.gnu.org. Please visit there. Sourceforge is generally depreciated except for mailing lists, but updated releases will be posted here from time to time.