Search Results for "audio"

jfPhone is a VoIP/SIP SoftPhone for Desktops.

Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players.

Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

...The Bria Desktop allows third-party applications to control Bria for Windows softphone clients. By leveraging the Application API, third-party applications can perform commands such as starting an audio or video call, answering a call or placing a call on hold. CounterPath is now encouraging third-party-application developers to integrate their applications with Bria. Suggested applications range from simple functions such as enabling click-to-call from web pages, to integration with sophisticated Call Center, Customer Relationship Management (CRM) and Health Care applications. ...

This is a P2P VoIP system. It support audio and video chat through kademlia-based dht overlay. The p2p module use Kademlia protocols; Audio codec is G.729. An simple Introduction here: http://www.needdo.com/2007/09/ppphonekademliap2p-voip/ . My Personal blog is http://www.idolf.cn and my email is dolfcao@gmail.com .

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

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

Simple testing tool to generate RTP data packets and send it via netwok interface or save into pcap file. Primarily intended for use with SIPp application to test speech quality with different codecs.

PJSGUA is an Open Source SIP UAC based on PJSUA reference UAC from PJSIP project

Skype Audio Player allows playing audio files (mp3, wav, wma) during Skype calls to both parties. It is a simple tool without luxurious functions, useful for those who study foreign languages using Skype and need to play sample dialogs or audio tests.

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.

PcToPcChat is a new idea on video and audio chat witch is better than Skype, Yahoo ... It is a VOIP application (a video phone). The image and voice is synchronized.

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.

This project develops video conferencing and VoIP applications for Mac OS X. XMeeting is based on the work of the OpenH323 project which provides the libraries for H.323 and SIP support.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

Caller ID is a callerid management application to keep track of calls both locally and remotely. Offers many features including voice mail, real-time notification over the internet while the phone is still ringing. Send using MSM, Email, FTP etc.

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.

A patent-free audio codec designed especially for voice (unlike Vorbis which targets general audio) signals and providing good narrowband and wideband quality. This project is complementary to the Ogg Vorbis codec.

Scx-tools is a set of utilities for Skype VoIP client. The project goal is to provide several useful tools for such task as Skype calls recording and management. Currently only the call recorder (version 0.1.1) is available.

A C# PocketPC project for using your favourite SIP-provider with a Wifi connection for making inexpensive calls on your pocket PC. Currently tested with Asterisk server, FreeWorldDialup, Voipbuster.

vox-attendant is a telephony voice enabled virtual receptionist designed to work with VoiceXML and includes a distribution for the Voxeo Prophecy platform. It provides a web based interface for managing the directory.

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.

The man goal of Dr.Ta/MOD project is to create a brand new communication in vehicle. Under multi-network environment, Dr.ta/Mod use SCTP multi-homing feature to process multi-path association. Please read more in our website http://mod.maple.tw

SIPNet is a p2p-sip framework, it support IM,audio,video based on p2p-sip protocols, and support file sharing like other p2p programs such as bittorrent. All SIPNet clients join to form a P2P overlay net and find each other using DHT algorithm.

Enable Linux firewall to support connection tracking and NAT of H.323 protocol. It supports RAS, Fast Start, H.245 Tunnelling, Call Forwarding, Signal Proxy/Softswitch, RTP/RTCP and T.120 based audio, video, fax, chat, whiteboard, file transfer, etc.