Search Results for "sofia-sip" - Page 2
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Showing 323 open source projects for "sofia-sip"
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Jitsi Desktop
Jitsi is an audio/video and chat communicator that supports protocols
Jitsi Desktop is a free open-source audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features. Jitsi Desktop is the heritage of Jitsi Meet. While some components are still used in e.g. Jigasi, the project is not actively developed anymore. Improvements, bugfixes and builds are entirely based on community contributions. -
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Calis-p
Estimates delta13C of species in a microbiome from proteome data
Calis-p (The CALgary approach to ISotopes in proteomics) is a java application to estimate isotopic composition (e.g. delta13C or delta15N) of individual species in a microbial community from a proteomic dataset. Calis-p 2.0 handles both natural isotope abundances and data from labelling experiments such as stable isotope probing (SIP). It requires a mzIdent (or target spectrum match) and mzML files as the input and requires about 1 min per mzML file with 10 threads and needs <10 Gb of RAM... -
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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Bt_proteindatabase
Specific Bacillus thuringiensis protien Blast database
A specific protein database of insecticidal genes (Cry, Vip, Sip, Mtx-like, Bin-like and other protins) from the Bacillus thuringiensis toxin nomenclature. -
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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go -
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Asterisk Channels Live
This is a PBX dashboard showing the peer,queue,agent at real time
http://www.astchannelslive.com/sourceforge.php This is a dashboard work with Asterisk(FreePBX) any other installation: 1. Showing the peer status 2. queue,agent at real time 3. Hangup, Spy,.... -
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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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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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driveinfo
A macOS command which provides partition information for GPT drives.
... can no longer read the GPT on an operating system boot drive, unless System Integrity Protection (SIP) is disabled. The driveinfo command does not require disabling SIP and can produce the same basic tabular output as the gpt command. -
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AutoBridge
Home Agent Solution for Avaya by AutoBridge
The AutoBridge enables Home Agent solution for Avaya contact center. The tool monitors a pool of Avaya extensions in the contact center, it bridges Avaya and Asterisk extensions together whenever any of the Avaya extension has incoming or outgoing call. This enables home agent who is stayed at home can operate his/her CRM application via a VPN connection and speaks to his/her customer using an Internet SIP phone. -
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Protocol plugin for Office 365/Lync/OCS
for Adium, Pidgin, Miranda and Telepathy IM Framework
Please make sure to read this page: http://sourceforge.net/p/sipe/wiki/Frequently%20Asked%20Questions/ A third-party Pidgin/Adium/Miranda/Telepathy plugin for the extended version of SIP/SIMPLE used by various products: * Skype for Business * Microsoft Office 365 * Microsoft Business Productivity Online Suite (BPOS) * Microsoft Lync Server * Microsoft Office Communications Server (OCS 2007/2007 R2) * Microsoft Live Communications Server (LCS 2003/2005) -
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TeslaPSTN
Use Tesla API with Asterisk
A simple shell script that allows you to use the Tesla API through Asterisk via the Public Switched Telephone Network (or through any other channel that Asterisk supports, such as SIP). -
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TeeBX
Asterisk based VoIP pbx appliance
TeeBX wants to be a VoIP appliance with a user interface to easy manage a communication platform based on Asterisk (forked from askozia pbx). Website, support, documentation (coming soon): http://www.teebx.com/ -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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Sometimes in the process of debugging SIP UAs you are trying to simulate a scenario for which all you have is a packet trace from the user. You can use this program to convert the SIP packets from the pcap trace to sipp xml files, and then use sipp with the resulted scenario to simulate the test case. Note: only supports traces with SIP over UDP See "pcap2sipp doc" for more information
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Xplico
Xplico is a Network Forensic Analysis Tool (NFAT)
Xplico is a Network Forensic Analysis Tool (NFAT). The goal of Xplico is extract from an internet traffic capture the applications data contained. For example, from a pcap file Xplico extracts each email (POP, IMAP, and SMTP protocols), all HTTP contents, each VoIP call (SIP, MGCP, MEGACO, RTP), IRC, WhatsApp... Xplico is able to classify more than 140 (application) protocols. Xplico cam be used as sniffer-decoder if used in "live mode" or in conjunction with netsniff-ng. Xplico is used... -
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SipTunnel is used to tunnel SIP UDP datagrams (and also RTP datagrams) through NAT using TCP protocol. It also can be used for many other purposes.
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Vodia PBX
A Better PBX For Your Business
Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license. -
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Accent Tool Suite
Accent interfaces and controls systems through goals and policies
Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances... -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call... -
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Andspoilt
Run interactive android exploits in linux.
Andspoilt is a command line user interface designed to easily exploit android devices. Run interactive android exploits in linux by giving the users easy interface to exploit android devices Uses an intergration with Metaspoilt Framework by giving the user an easy interface to create payloads and launch android exploits. Current Additional feature is a simple web server for file distribution. -
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TOFFEE-DataCenter
Linux Server and Data-Center WAN Optimization
TOFFEE-DataCenter is the new TOFFEE variant meant for Data-Center, Server, HPC, Load Balancing, Cluster/Cloud Computing and SDN deployments. With TOFFEE-DataCenter (and or TOFFEE), OEMs and other equipment vendors can make use of this platform/stack, integrate, port and build WAN Optimization devices/appliances on to their existing product range. Or build a dedicated WAN Optimization hardware product series via the same. -
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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StarterWareFree for AM335X
Community-driven variant of TI's bare metal development support tools
This is the community driven variant of StarterWare(tm) 2.0.x by Texas Instruments, a collection of example source codes, headers, and basic implementations for AM335X-based hardware (like the BeagleBone Black, the PocketBeagle or the OSD335x SiP). The aim of this project is to add features and functionalities faster than TI is able to do that and to get a high-quality package for easy bare metal programming on such Sitara-based devices.