Search Results for "http c tiny"
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Showing 29 open source projects for "http c tiny"
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Ship Agents FasterGemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
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Free Telephony Project
Free software and hardware for telephony
Legacy SVN repositories for projects from David Rowe http://rowetel.com Mailing lists for Codec 2 and SM1000. -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on... -
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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/
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GetSmart is a download manager that supports Multi-Connection downloads. With Multi-Mirror search it can download different parts from these mirror servers simultaneously. It can 'act as proxy' to serve multiple clients. It can also works as a Daemon.
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Custom VMs From 1 to 96 vCPUs With 99.95% UptimeLive migration and automatic failover keep workloads online through maintenance. One free e2-micro VM every month.
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wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .
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This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated:... -
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for... -
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Seagull is a multi-protocol traffic generator. Especially targeted towards IMS, Seagull supports Diameter (RFC3588 and all applications) over TCP/SCTP and IPv4/IPv6 , TCAP (over SS7 or Sigtran), XCAP over HTTP and Radius.
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sigtran-smsc
PoC: Short Message Service Center (SMSC) over SIGTRAN
SIGTRAN based Short Message Center (SMSC) - proof of concept This is home of my thesis project which had goal of doing open source based Short Message Service Center to be connected into real mobile network (2G, 3G, ...). As being developed in very limited schedule (bachelor thesis work) it was limited to have very basic functionality: receiving message and deliver it immediately as-is to the receiver. There is no message caching nor different message charset support. It uses... -
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The ASN.1 Compiler
Go to github.com/vlm/asn1c for the latest version.
This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.
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The OpenSIPStack Library is an implementation of the Session Initiation Protocol as described in RFC 3261. Applications: * OpenSBC (B2BUA) * OSSPhone (Softphone) Source code is available at http://www.opensipstack.org.
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Using the new C# Dll from swig portability of Openobex/ObexFtp to write a nice GUI able to transfer file between two devices with obex protocol. More Info’s here: http://cs-obexftp.wiki.sourceforge.net/
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La Suite Nemesis Research & Design 1.x (NemesisRD) es un conjunto de librerías multipropósito, que ofrece gran cantidad de clases y patrones implementados en C++, que tienen como objetivo facilitar la realización de las tareas más habituales a la hor
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This an open source project for Paliha P750 and similar corded phones http://www.paliha.ru (only Russian language). The original project "Teleport-Paliha" was written for Windows OS, but I want the same on my Linux!
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HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت
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OpenSMSC is ETSI GSM based realtime, distributed, highly available Short Message Service Center. It transfers short messages between mobile phones(MO and MT) over SS7 network and to applications(ESME) using open standard protocols like SMPP, HTTP & SOAP
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Dgvchannel é um driver que permite a utilização das placas DigiVoice com o Asterisk(c). A Digivoice tem uma placa E1 (1 ou 2 spans) e uma placa FXO (4 ou 8 canais) que são suportadas por este channel. Acesse http://www.digivoice.com.br
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Skinny (SCCP) Channel Driver for the Asterisk Open Source VOIP Platform. This project has been superseded by the chan-sccp-b project at sourceforce !! => http://chan-sccp-b.sourceforge.net
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Time of day service over telephone using Voicent Gateway, a VoiceXML gateway that specially designed for voice modems. A Free version is available for download at http://www.voicent.com/download. Sample code for interactive telephony applications.
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BeepagePHP provides a PHP-based web client to the beepage client/server paging system. Beepage is a Unix-based, Internet-aware text paging system that implements the Telocator Alphanumeric Protocol (TAP). Source code is hosted at http://git.io/rgikgg
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