Search Results for "rtp-text"

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/RTCP/T.38/udptl protocols. ...

The VoIP voice-to-text software for Issabel is an intelligent, AI-based solution that converts calls into accurate Persian text. After each call, the audio file is sent to the GPT-4O AI engine, producing editable transcripts. The software also provides AI-powered call analysis, extracting key points, customer requests, satisfaction levels, and sensitive topics, all stored in the database.

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/

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 Centers and Contact Centers for Quality monitoring (QM) purposes.

...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.

tui is a user interface for mobile phones with keypads. It runs on text console or under nano-X, and communicates with modem part with AT commands. It runs well on Siemens SX1 running linux, and allows you to call from it and answer calls.

A bash shell script which will read active call destinations in E.164 format from a source (such as Asterisk), then run the called number through some plain text and MySQL databases to find the best latitude/longitude match. It then creates a .html file with Google Maps API JavaScript code which will display a single call origination point and the active destinations (calls) on a Google Maps world map.

mmsdec is an application developed in C which can be uses to decode MMS message (Multimedia Messaging Service). With this application it is possible to extract text message, photos and videos from a raw mms message.

"Way to Sms" is a desktop application written in python to send free text messages all over India through Way2Sms gateway

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.

...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.

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

app_swift provides an interface into the Cepstral TTS libraries to stream text-to-speech audio from the Asterisk PBX without creating files.

Use this when attachments aren't offered with your email service. (i.e. Proxy Mail). Simply Supply a link to download the 11KB Audio Player and Copy/Paste your Message into the Email. See Website For Demo. http://audiotext.sourceforge.net

simple sms gateway written in tcl/expect, using mysql as datasource for the text messages.

A modularly designed system providing command/trigger functionality and response mechanism via SMS text messages

Frosty is a php based application that will allow you to send sms messages to a wide variety of people. Frosty uses email-to-text and google voice and can send a single message up to 100 times to one recipient or to 10000 unknown recipients!

Quintum Tenor VoIP Gateway Call Detail Record (CDR) Advanced Server. This CDR Server can collect Call Detail Records (CDRs) from multiple Tenors. The data is collected simultaneously and continuously. Supported MySQL and file storages.

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.

A simple SMS gateway for Linux written in C. Capable of sending and receiving SMS text messages like a mobile phone. An additional java-package is provided for easy communication with the integrated TCP-server.

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.

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

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.

U9input designed to allow text input with a restricted number of keys. User defined keys relate to a set of letter and the program then presents possible words that match. It can be configured to allows it to emulate T9.

eSpeak text-to-speech module for Asterisk. This provides the "espeak" dialplan application, which allows you to use the eSpeak TTS Engine as a speech synthesizer in Asterisk.