Search Results for "sofia-sip"
Sort By:
35 projects for "sofia-sip" with 2 filters applied:
-
Our Free Plans just got better! | Auth0You asked, we delivered! Auth0 is excited to expand our Free and Paid plans to include more options so you can focus on building, deploying, and scaling applications without having to worry about your security. Auth0 now, thank yourself later.
-
$300 Free Credits to Build on Google CloudStart your next project with $300 in free Google Cloud credit. Spin up VMs, run containers, query petabytes in BigQuery, or build agents with Gemini Enterprise Agent Platform. Once your credits are used, keep building with 20+ always-free tier products including Compute Engine, Cloud Storage, GKE, and Cloud Run functions. No commitment required—just sign up and start building.
-
1
VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. ... -
2
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.
-
3
Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
4
jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
Build Securely on Azure with Proven FrameworksMoving to the cloud brings new challenges. How can you manage a larger attack surface while ensuring great network performance? Turn to Fortinet’s Tested Reference Architectures, blueprints for designing and securing cloud environments built by cybersecurity experts. Learn more and explore use cases in this white paper.
-
5
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/
-
6
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
-
7
Taki
SIP softphone
Cross-platform SIP softphone -
8
Callflow Sequence Diagram Generator
Callflow Sequence Diagram Generator
The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic -
9
jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net -
Build Agents and Models on One PlatformGemini Enterprise Agent Platform is Google Cloud's comprehensive platform for developers to build, scale, govern, and optimize agents and models. Choose from Google's most advanced models and third-party models like Anthropic's Claude Model Family.
-
10
SIP Data Filter (SiDaFir)
Simple and efficient tool for SIP trace filtering
...Aim of this project is to offer a simple and effective, yet well-configurable, tool allowing for SIP trace filtering - the SIP Data Filter (SiDaFir). -
11
Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
-
12
Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
-
13
baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
14
Sippy B2BUA and RFC3261 SIP Stack
This project has been relocated to http://github.com/sippy/rtpproxy h
This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua -
15
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 example, HTTP servers). ... -
16
Artemisa is a honeypot for VoIP (SIP) networks. It is designed to connect to a VoIP domain as a user-agent backend in order to detect malicious activity at an early stage and also adjust the policies of the enterprise in real-time.
-
17
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
-
18
LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git
-
19
CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
20
SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.
-
21
SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi
-
22
SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
-
23
SIP.NET is SIP stack .NET library written in C#. SIP message parser realized on Deterministic Finite-state Machine (DFA parse all SIP message for one pass). DFA generates automatically from Augmented Backus–Naur Form from RFC3261.
-
24
OpenBTS
GSM L1-L3 stack with SIP network interfaces
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface. -
25
Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.