Einfache Projektliste Software-Karte

Internet-Telefon
166 Projekte im Ergebnis
Letztes Update: 2011-05-22 14:01

Sofia-SIP

Sofia-SIP is a SIP user agent library, compliant
with the IETF RFC3261 specification. It can be
used as a building block for SIP client software
for uses such as VoIP, IM, and many other
real-time and person-to-person communication
services. The primary target platform is
GNU/Linux. Sofia-SIP is based on a SIP stack
developed at the Nokia Research Center.

(Machine Translation)
Letztes Update: 2019-06-14 19:04

Elastix

Elastixは、AsteriskベースのPBXに使いやすいインターフェースを付加するためにベストなツール群を統合化したアプライアンスソフトウェアです。また、オープンソースのテレフォニーのためのベストなソフトウェアパッケージとするために、独自のユーティリティの設定も追加されています。

Letztes Update: 2003-07-31 05:33

Robust Audio Tool

RAT is an RTP audio conferencing and streaming application that allows users to particpate in point-to-point and multi-point audio conferences over the Internet. Developing features include multiple sampling rates, mono/stereo, loss concealment, and IPv4/IPv6 support, and both versions feature encryption and run on a range of platforms.

Letztes Update: 2012-03-19 01:33

Asterisell

Asterisell is a Web application for rating, showing to customers, and billing of Asterisk VoIP calls.

(Machine Translation)
Letztes Update: 2009-05-24 07:00

sip-redirect

sip-redirect is a tiny SIP redirect server. It supports IPv4 and IPv6, but the IPv6 support is optional. The RFC 3261 was the base for this simple and very configurable implementation. There is neither TCP nor multicast support programmed in.

Letztes Update: 2011-02-15 13:56

CELT audio codec

CELT (Constrained Energy Lapped Transform) is an
ultra-low delay audio codec designed for realtime
transmission of high quality speech and audio.
This is meant to close the gap between traditional
speech codecs (such as Speex) and traditional
audio codecs (such as Vorbis).

(Machine Translation)
Letztes Update: 2005-04-12 09:58

Cutlass

Cutlass is a cross-platform system for secure peer to peer communication, oriented towards small groups. It provides VoIP, IM and file transfer. A major design goal is to be easy to use, even by non-security conscious users.

Letztes Update: 2011-12-26 22:34

mediastreamer

Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.

Letztes Update: 2009-01-19 18:28

PJSIP and PJMEDIA

pjsip is a multimedia communication library based on the SIP protocol. It is integrated with a rich media and a NAT traversal library supporting the ICE protocol. It is very portable and has a small footprint for embedded use.

Letztes Update: 2012-01-08 00:16

SIPFwd

The SIP forwarding daemon (implemented as a stateless SIP proxy) allows you to seamlessly forward SIP requests to other SIP servers. Its main purpose is to enable users to use their own domain name in SIP URIs without the hassle of having to run a full-blown SIP server (by forwarding SIP requests to third-party SIP servers). Configuration information is stored in an SQLite database, and low resource consumption is a main priority for the project.

Letztes Update: 2002-03-03 09:46

jAugment

jAugment is a JINI-based, network-aware framework and set of initial applications for wearable computers and other computers with uncommon input/output-devices. It supports user interfaces from text-only to 3D.

(Machine Translation)
Letztes Update: 2007-05-05 05:35

libsrtp

libsrtpライブラリセキュアRTPには、Secureのリアルタイム転送プロトコルの実装です。 RTPのボイスオーバーIP(VoIP)だけでなく、オーディオやビデオのストリーミングに使用されます。SRTPをconfidentialtiyと認証を追加します。

(Machine Translation)
Letztes Update: 2005-08-30 11:27

Arsenal

Arsenal is a realtime collaboration (RTC) and conferencing platform. It supports presence, instant messaging, filesharing, voice conferencing, persistent sessions for sharing whiteboard, web browser, images, and group chat.

(Machine Translation)
Letztes Update: 2006-06-08 09:18

Star Asterisk API

Star Asterisk API is a high performance API that
connects to the manager interface of Asterisk or
to AstManProxy. It is easy to use,
object-oriented, and easy to extend to suit your
particular requirements.

(Machine Translation)
Letztes Update: 2008-01-30 17:26

I Hear U

IHU is a Voice over IP (VoIP) application that creates an audio stream between two computers easily and with the minimal network traffic. It works without an intervening server, can use either UDP or TCP for transport, can encrypt the data stream, and can work without a GUI. It uses the Speex codec to help reduce bandwidth consumption. It supports ALSA and JACK for audio service.