Search Results for "http-client"

Note: for binaries and installation instruction please visit official website. http://ictfax.org/ ICTFax is Open Source Fax Server software to send and receive Fax over Internet & Emails developed over Freeswitch, ICTCore and Angular Framework , It works with T.38 and G.711 pass through and feature email to fax, fax to email, web to fax, fax to web and also support fax machines to send and recieve Fax using ATA.

...Additional features are in development. This is a mirror of the code on https://github.com/jselbie/stunserver More details on the project's website: http://www.stunprotocol.org

...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/

TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.

...REQUIREMENTS: - Java 7 or higher for Windows. - Java 6 or higher for Mac. Java 7 recommended. - A virtual audio cable is required for use with VOIPs: For Windows users I recommend the VB-Cable driver (http://vb-audio.pagesperso-orange.fr/Cable/index.htm). Mac users can use SoundFlower.

Note: For binary packages and updated installation instructions please visit this link http://www.ictinnovations.com/content/ictinvoice-complete-multitenant-hosted-pbx-solution-service-providers ICTInvoice is multi tenant Invoicing module of Elastix PBX. It is complete invoice management solution for multiple companies. It automatically create pdf invoices and emails to Elastix users on monthly basis along with their calls detail records ICTInvoice Main Features Multitenant Create invoices for Multiple campanies with branding Manual as well as Automatic invoice generation on fix date Auto Emailing invoices to users on monthly basis Inbound Billing Outbound Billing Extension / User wise reporting Summrize reporting Detail Reporting For more information , please visit http://www.ictinnovations.com

############################################### #--- CYBER-WRT by Christian Zeler / Comdif Telecom ---# #----------------- http://cyberhotel.comdif.com ------------------# ############################################## You can find 2015 news Cyber-hotel versions on http://cyberhotel.comdif.com including cyber-hotel on freepbx - Cyber-hotel decktop. . Cyber-wrt is a complet Asterisk IPBX and Hotspot system embedded in DD-Wrt router. Last version is build for ASUS RT-AC56U router It is working on Debian with asterisk pbx and original captive portal system to offer a reliable Hotspot system. ...

This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua

...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 example, HTTP servers). Our aim is to promote analysis of real SIP-server behavior. Therefore, we present SiAnTo, an extended anonymization technique that substitutes session-participant information with matching, but nondescript, labels. This allows for SIP traces to be publicly shared, while keeping interesting traffic-session properties intact.

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

node (nodejs) non-blocking RADIUS client wrapper

VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K

FreePBX module for ENUM lookup API at enumplus.org.

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 simple lightweight instant messaging and VoIP client wirtten in Java. It implements the SIP protocol as described in RFC 3621 and other related RFCs and implementation requierements.

VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z

LibDreamSpeak is an example implementation of the TeamSpeak2 protocol (client-side) in Java. Some day it may be used as base for alternative TeamSpeak2 (mobile?) clients.

Queue-Tip has moved to Rubyforge - please go to http://queue-tip.rubyforge.org

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.

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.

SipManager is a full-feature IP-Centrex PBX web app for hosted/multi-client PBX based on Asterisk and OpenSer. VoIP/ToIP platform (PBX, Fax, workflows, etc.). User-friendly web interface. Realtime architecture with SQL database. ASP. http://www.ovvoe.com

...By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.

Cross-platform Java Teamspeak client

AS3 skype API for the Skype. Flash has access to the system Webcam, Audio and flash 9+ has all the capabilities to build a Socket communications. So we can have a Flash based skype client on the web. This can be a widget for instant chat & voice call.