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/

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.

...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.

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.

Adds a Right Click context menu to your browser to call number from Lync. Highlight a piece of text in IE/FF13/Chrome and the script will strip out non numerics and pass to Lync via tel: handler and Lync can normalise the number.

A simple status bar item that sets your Skype mood message to the song that is playing on iTunes, VLC or Spotify.

OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list

...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

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

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 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.