Einfache Projektliste Software-Karte

395 Projekte im Ergebnis
Letztes Update: 2007-04-27 08:34

Sipp

Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.

(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: 2006-03-01 07:29

SCMxx

SCMxx is a console program that allows you to
exchange certain types of data with mobile phones
made by Siemens. Some of the data types that can
be exchanged are logos, ring tones, vCalendars,
vCards, phonebook entries, and SMS messages. It
works with the following phones: S25, C35i, M35i,
S35i, ME45, S45, SL45, M50, and probably some
others, too. It basically uses the AT command set
published by Siemens (with some other additional
resources).

(Machine Translation)
Letztes Update: 2002-05-22 22:34

CPhone

CPhone is a cross-platform GUI for the OpenH323 VOIP libraries.

Letztes Update: 2009-05-13 14:22

Gnu Gatekeeper ACD

The GnuGk ACD does automatic call distribution for the Gnu Gatekeeper. It is the first step to a VOIP call-center solution based on GnuGk. Incoming calls are routed to agents based on different distribution algorithms.

(Machine Translation)
Letztes Update: 2002-05-09 09:02

gsmlib

GSMLIB is a library to access GSM mobile phones through GSM modems. Features include: modification of phonebooks stored in the mobile phone or on the SIM card, reading and writing of SMS messages stored in the mobile phone, sending and reception of SMS messages. Additionally, some simple command line programs are provided to use these features.

(Machine Translation)
Letztes Update: 2007-07-03 11:49

Asterisk Soap API

Asterisk-soap is an extensible SOAP (Web services)
API with the aim to support all administration
features available in the Asterisk VoIP server.
Its primary goal is to provide a framework between
Asterisk and multi-platform frontends.

(Machine Translation)
Letztes Update: 2012-04-22 12:29

restund

restund is a modular STUN/TURN server that is designed around the principle of a lightweight core and server modules that extend its functionality. Both UDP and TCP are supported, along with IPv6 and IPv4. Supported modules include STUN, TURN, MySQL database, syslog, and status monitoring.

Letztes Update: 2010-10-19 14:50

QuteCom

QuteCom (formerly WengoPhone) is a multi-platform
VoIP client. The GUI is Qt-based, and the
Video-over-IP engine is based on the eXosip, oSIP,
oRTP, ffmpeg, and libgaim projects. The eXosip
module is extended by a phApi module, which
implements a high-level, easy-to-use call control
API. It supports PC-to-PC voice, video, and chat.
One can use a standard SIP service provider such
as Wengo to be assigned an incoming number, make
calls to PSTN/cell phones, get voice mail, and
more. In addition to SIP/SIMPLE QuteCom provides
IM functionality using libgaim, so it is
compatible with all protocols supported by libgaim.

(Machine Translation)
Letztes Update: 2004-12-21 10:19

asterisk-oh323

asterisk-oh323 adds H.323 support to the ASTERISK soft PBX. It does this
by interfacing the OpenH323 library to ASTERISK through a loadable
module. The package provides the channel driver as well as a wrapper in
a shared library form. It is able to initiate and receive calls to and
from H.323 endpoints, and has been successfully tested with the H.323
terminals on the OpenH323 site (ohphone, openphone), GnomeMeeting,
Microsoft NetMeeting, Cisco routers, H.323 Snom phones, and other hardware and software H.323 IP phones.

(Machine Translation)
Letztes Update: 2001-04-13 08:59

Open SS7

SS7 コア プロトコル、MTP、SCCP、ISUP、および TCAP のオープン実装です。

(Machine Translation)
Letztes Update: 2009-02-01 16:53

Jiplet Container

Jiplet Container (Java SIP Servlet) is a servlet-like development and runtime environment for SIP applications. The SIP protocol is widely used for voice services over IP networks. This product enables developers to create server-side SIP applications using a component-based model similar to that envisioned by the J2EE architecture. The Jiplet container runs as a standalone server as well as a JBOSS service.

(Machine Translation)
Letztes Update: 2019-03-08 00:16

OpenSIPS/OpenSER-a versatile SIP Server

OpenSIPS(旧OpenSER)は、GPLライセンスの元に実装している多機能なSIPサーバで、プロフェッショナルなSIPサーバプラットフォームで使用されるハイレベルな技術的ソリューション(パフォーマンス、セキュリティ及び品質)を提供することを目的としています。

Databank Umgebung: Other API
Entwicklungsstatus: 6 - Reifen
Natürliche Sprache: English
Programmiersprache: C
Benutzerschnittstelle: No Input/Output (Daemon)
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: 2013-06-22 22:15

pyst

Pyst consists of a set of interfaces and libraries to allow programming of Asterisk from Python. The library currently supports AGI, AMI, and the parsing of Asterisk configuration files. The library also includes debugging facilities for AGI.

(Machine Translation)