Простой sip сервер для windows

Table of Content

SIP stands for Session Initiation Protocol, which is a signaling protocol for initiating, maintaining, and terminating communication sessions that include voice, video, and messaging applications.

SIP clients

SIP clients is an internet telephony software, that allows you to make voice and video calls over the internet using VoIP. Android provides an API that supports the Session Initiation Protocol (SIP). This lets you add SIP-based internet telephony features to your applications.

What is the difference between SIP and VoIP?

VoIP, or Voice over Internet Protocol, is a technologies that enables voice to be sent over the Internet, like Skype, and many other services.

On the other hand, SIP (Session Initiation Protocol), is a protocol that can be used to set up and take down VoIP calls, and can also be used to send multimedia messages over the Internet using PCs and mobile devices.

Open source SIP servers

SIP server is an essential tool that facilitates internet-based telephony. It connects your company’s IP PBX to an internet telephony service provider (ITSP).

SIP open source servers allows you to create your own server with a low cost, unlike many commercial alternatives.

Here is our list:

1- OpenSIPS

OpenSIPS is a free open source SIP proxy/ server that supports voice, video, IM, presence, and other SIP extensions.

OpenSIPS team offers a LTS support for latest stable release, and it is available for Linux servers (Ubuntu, Debian, Fedora, openSUSE, RedHat, and CentOS).

It is a multi-functional, multipurpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions. Its features also include Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others features.

OpenSIPS Features

OpenSIPS has to offer many important and interesting features. To mention some of the most important ones:

  • SIP registrar server
  • SIP router / proxy (lcr, dynamic routing, dialplan features)
  • SIP redirect server
  • SIP presence agent
  • SIP back-to-back User Agent
  • SIP IM server (chat and end-2-end IM)
  • SIP to SMS gateway (bidirectional)
  • SIP to XMPP gateway for presence and IM (bidirectional)
  • SIP load-balancer or dispatcher
  • SIP front end for gateways/asterisk
  • SIP NAT traversal unit
  • SIP application server

2- Kamailio

Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPLv2+, able to handle thousands of call setups per second. It is a popular choice for many companies to handle large SIP and VoIP communication.

Kamailio can be used to build large platforms for VoIP and real-time communications – presence, WebRTC, Instant messaging and other applications. Moreover, it can be easily used for scaling up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.

The Kamailio SIP server is designed for scalability, targeting large deployments (e.g. for IP telephony operators or carriers, which have a large subscriber base or route a big volume of calls). However, it can also be used in enterprises or for personal needs to provide VoIP, Instant Messaging and Presence.

Kamailio project has a rich documentation that includes a long instruction set on how to install, configure, integrate, and use.

The development was started back in 2001 by Fraunhofer Fokus, a research institute in Berlin, Germany.

Kamailio can be installed on Debian, Ubuntu servers, which are officially supported by the development team. It can also be installed on any server using Docker and Ansible.

Kamailio is released under the GPLv2 License.

Kamailio SIP Server

Welcome To Kamailio — The Open Source SIP Server Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPLv2+, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and realtime communications — presence, WebR…

The Kamailio SIP Server Project | The Open Source SIP Server

3- Drachtio

Drachtio is a SIP server for developers that help them to build SIP apps simply as building web apps. It has a core framework which called Drachtio Signaling Resource framework (drachtio-srf), the Node.js framework for SIP Server applications.

So, if you want to build web apps using JavaScript or TypeScript, then Drachtio is your choice.

Drachtio is released under the MIT License.

GitHub — drachtio/drachtio-server: A SIP call processing server that can be controlled via nodejs applications

A SIP call processing server that can be controlled via nodejs applications — GitHub — drachtio/drachtio-server: A SIP call processing server that can be controlled via nodejs applications

GitHubdrachtio

4- Asterisk

It would be unfair to finish this post without talking about Asterisk, which is a complete-integrated solution for for internet-based telephony. It offers a LTS (Long Term Support) stable edition, that is easy to install and configure.

GitHub — asterisk/asterisk: The official Asterisk Project repository.

The official Asterisk Project repository. Contribute to asterisk/asterisk development by creating an account on GitHub.

GitHubasterisk

5- Sip Server

Sip Server is a simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms.

GitHub — BarGabriel/SipServer: A simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms.

A simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms. — GitHub — BarGabriel/SipServer: A simple SIP server (proxy) for handling VoIP calls bas…

GitHubBarGabriel

6- LibreSBC

LibreSBC is an open-source Session Border Controller to provide robust security, simplified interoperability, advanced session management, high performance, scale of carrier-grade and reliability for voice over IP (VoIP) infrastructures.

LibreSBC designed to typically deployed at the network edge, the demarcation points (borders) among networks/environments.

GitHub — hnimminh/libresbc: An open source Session Border Controller, The SBC you dream about 🌟 LibreSBC will help you save thousands of dollars.

An open source Session Border Controller, The SBC you dream about 🌟 LibreSBC will help you save thousands of dollars. — GitHub — hnimminh/libresbc: An open source Session Border Controller, The SBC…

GitHubhnimminh

7- SIPp

SIPp is a free Open Source test tool / traffic generator for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC and UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It can also read custom XML scenario files describing from very simple to complex call flows.

It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.

Other advanced features include support of IPv6, TLS, SCTP, SIP authentication, conditional scenarios, UDP retransmissions, error robustness (call timeout, protocol defense), call specific variable. Moreover, it supports Posix regular expression to extract and re-inject any protocol fields, custom actions (log, system command exec, call stop) on message receive, field injection from external CSV file to emulate live users.

SIPp can also send media (RTP) traffic through RTP echo and RTP / pcap replay. Media can be audio or video.

GitHub — SIPp/sipp: The SIPp testing tool

The SIPp testing tool. Contribute to SIPp/sipp development by creating an account on GitHub.

GitHubSIPp

8- Hermes

Hermes is a modern SIP server framework for building real-time SIP apps. Hermes will substitute old legacy SipServlet. It is based on reactive manifesto.

Hermes is meant for Java developers, and it is a FLOSS (Free Libre Open Source Software) under the GNU Lesser General Public License.

GitHub — owen-q/Hermes: Modern Java SIP Framework based on reactive

Modern Java SIP Framework based on reactive. Contribute to owen-q/Hermes development by creating an account on GitHub.

GitHubowen-q


If you know of any other open source SIP server that we missed, let us know.

You may check our updated list:

22 Open-source Free VoIP and Sip Servers

A VoIP (Voice over Internet Protocol) server is a computer system that enables voice communications over the internet. It converts analog audio signals into digital data packets and transmits them over the internet. SIP (Session Initiation Protocol) is a signaling protocol used for initiating, maint…

MEDevel.comHamza Mousa

A SIP server or Session Initiation Protocol server is a must require tool if you want to start a business regarding Voice over IP telephony. SIP is open-source server software that comes to hooks up computer programs or libraries. And it’s the key element of an IP PBX and primarily deals with handling all SIP calls in the network. Hence, you’ll learn very different servers’ names with their pros and best features that will deliver from this post.

Before knowing the best free sip server software, let’s have a quick chat about sip server. The complete form of SIP is the Session Initiation Protocol. And it’s a TCP/IP-based web protocol that uses for connecting and controlling customer contact. SIP comes with VoIP (Voice over IP) telephony to set up links for phone calls, and its main features are defined in SIP RFC3261.

But a Session Initiation Protocol server is also familiar as a SIP proxy. It is responsible for soliciting requests from user agents to make and stop calls. And this server empowers you to control call cohesions in VoIP solutions. So, you can tell, this server can:

  1. To set up a relation between countless endpoints.
  2. Using the SDP protocol to start the media parameters for the endpoint
  3. Change and revise the parameters during the session.
  4. Restore one specific endpoint with another or a unique endpoint
  5. Session consummation
best_open_source_sip_server_software_process

Now that you’ve learned what it is and how its server works, it’s time to get the best free open source sip server. And sad but true, if you search online about this topic, you’ll find many resources. But all are not as good as applicable. Hence, to make your job easier, I’ve developed a handy list that can provide your desired software.

Free SIP Server Software:

Table of Contents

So, without further delay, let’s jump deep to find out the best gems for you to use.

SIP Server Asterisk:

sip_server_asterisk

Asterisk is the materialization framework for PBX (Private Branch Exchange). Also, it’s the number #1 and free open source sip server software for making your private communication apps that Sangoma sponsors.

Mark Spencer of Digium built and designed it in 1999 initially for Linux. But it works perfectly on several operating systems such as macOSNetBSDSolarisFreeBSD, and OpenBSD.

Highlighted Features:

  1. Call Monitoring, Transfer, and Waiting
  2. Append Message, Blind Transfer, and Blacklists
  3. Making Real-time Communication Solutions
  4. Automatic Call Distributor Functionality
  5. Multi-protocol Solutions

On a quick note, it has 2,000,000 Downloads Yearly86,000 Community Mates, and 170 Countries with Installations. Also, it includes 1,000,000 Servers Globally and 1,300,000 Fresh Endpoints per year.

Open Source FreePBX Software:

best_open_source_sip_server_freepbx

FreePBX is the most famous free open source IP PBX tool worldwide. It gives you the freedom to create a phone system to suit your needs. And to create a scalable company phone system on any cost limits, it covers all the vital features.

Apart from that, it’s entirely free to download and exemplary easy to use. The global developer’s community ensures high compatibility and a customizable platform. Also, it’s a web-based graphical user interface (GUI) to manage Asterisk ( a Voice over IP) and telephony server.

Top Features:

  1. SmartOffice Access, Phones, and Appliances
  2. Session Initiation Protocol Trunking, VoIP Gateways, and Modules and Add-Ons
  3. Session Border Controllers
  4. Participate and Report Issues on Community Forums
  5. Bug Submissions

Above all, it has millions of installations globally with a very active blossoming base.

Most Popular Elastix Tool:

session_initiation_protocol_elastix

Does your communication need a PBX, Live chat, or Video? Want Video Conferencing, Presentation, and Teamwork tools in real-time? Okay, no problem, in this case, you can use Elastix. No need for any add-on fees or additional downloads, and you can easily install it on Windows, Linux, or Raspberry Pi as an on-premise solution.

Besides, it’ll help you with working remotely, sharing screens, no time limit for endless users, and so on. Also, it gives you the ability to flip web visits into leads and sales, leading call center features and narration, and much more.

Promising Features:

  1. App-free Online Conferencing
  2. Easy Install and Management
  3. Live Chat and Call with Site Callers
  4. Answer Fb Page Notes
  5. Offers Top class Client Service

And most importantly, it’s 24/7 available from your remote desktop or mobile gadget.

Award Winner Tool Vicidial:

free_sip_server_software_vicidial

Now, it’s time to introduce the most famous contact center solution worldwide named Vicidial. It’s an award-winner tool that grabs many awards regarding contact centers. And it found in 2007 through its real maker and initial developer, Matt Florell.

However, it can help you with single-agent call queuing, lead import web-based API, place emerging CallerID per operation or per list, and more. Also, it has a distinct Time-clock app to track user working time.

Best Features:

  1. Inbound email operating via agent web screen
  2. Call up to two different client numbers manually or automatically for the same lead.
  3. Run a campaign to auto-dial and forward live calls to public agents.
  4. Capability to auto record all calls
  5. Real-time campaign display screens and 3rd party blind call transfer

On top of that, it has more than 14,000 installations in 100 countries in 16 distinct languages.

Best Buddy Kamailio:

free_sip_server_software_kamailio

If you want to manage more than thousands of calls settings per sec, in this case, Kamailio will be your best buddy. It’s published under GPLv2+ and used to build vast platforms for Voice over IP (VoIP) and real-time communications. For example, you can be used this tool for instant chatting, WebRTC, and many various apps.

Also, it may best fit on climbing up SIP-to-PSTN gateways, media servers, or PBX systems. On the other hand, you can use it with limited resources and carrier-grade servers. And to provide high performance, it’s written on Unix/Linux systems, including architecture-specific optimizations.

Powerful Features:

  1. Asynchronous SCTP, UDP, and TCP
  2. Secure services through TLS for (voice, video, text) VoIP
  3. WebRTC (IPv4 and IPv6) on WebSocket support
  4. Routing fail-over, Least cost routing, and Load balancing
  5. Many backend systems support

In a word, its main aim is to be a combined habitat for its users to thrive on protected and scalable Session Initiation Protocol servers.

Telephony Network Solution GNU SIP Witch:

telephony_network_solution_gnu_sip_witch

The GNU SIP Witch designs to come forward to support telephony services network scaling instead of the excessively compute-bound solutions we use. It uses the Session Initiation Protocol to provide a protected peer-to-peer VoIP server. It comes as free software under the GNU General Public License (GPL) version 3 or later.

In addition to that, it’s constructed amazingly for macOS, BSD, Windows, and Linux, also for Android support. And it’s written in C++ and uses the uCommon programming language. So, it’s all about liberty to communicate and remove artificial fences and restrictions.

Robust Features:

  1. Presence information and text (messages)
  2. Supports encrypted calls and Enables NAT traversal
  3. Installed Ubuntu and Fedora directly
  4. Call Forwarding, Distribution, and Hold
  5. Self-organizing Peer-to-peer Telephone Network

On a serious note, it was used as an element of the GNU Free Call, making it a substitute for Skype.

Hey folks, now we’ve comes at the very end times of the listing. Did you find the answer to what you were looking for? If not, make a note to us, let’s consult, and we’ll make you flourish.

Or, want to know what about the future of VoIP technology for starting a business? In this scenario, you can read our other blog post, “How To Start A VoIP Business in 2022.”

Now come back to our topic…..!

Session Initiation Protocol proxy OpenSIPS:

session_initiation_protocol_proxy_opensips

OpenSIPS is a Session Initiation Protocol proxy or server for Voice, Video, IM, Presence. And it brings a potent and performant (Session Initiation Protocol RFC3261) Registrar, Location, or Redirect server. It also includes a springy and mighty scripting code for routing logic. It’s also the fastest server tool to deliver scalable explanations at an enterprise level.

On the other hand, it presents high-level technical solutions in professional platforms. For example, it gives technical solutions like Quality, Performance, and Security.

Key Features:

  1. Digest and IP Authentication
  2. Modular Architecture
  3. PERL Programming Interface
  4. UDP/TCP/TLS/SCTP Transport Layers Support
  5. SRV and NAPTR DNS Support

To clarify, it has a command interface through FIFO files and UNIX sockets.

Effective Server Tool Flexisip:

sip_server_tool_flexisip

Are you thinking of an effective and scalable Session Initiation Protocol proxy for topical routing of intercom calls? Okay, here is a server tool named Flexisip that will make your thought vanish. It’s ideal for integration within low-footprint embedded designs and massive cloud deployments.

Features:

  1. Real-time presence status and statistics
  2. Push notifications and Group chats
  3. Recognizing users of service
  4. Easy network deployment to Multicast DNS

Moreover, it’s easy to install, and you can use it for various purposes.

Conclusion:

Before picking a free sip server software from the above listing, please remember that a solo server can’t meet your every desire. Because every single sever has its distinct pros and cons. So, you’ve to choose wisely the best open source sip server that fills most of the requirements.

At last, I hope this post has enough info to grab the best server and make a path to start your business instantly. And if you like this blog post, please share it as much as possible with your friends and family, including social media.

Rest assured, this blog will never lead you astray but will support you from your side. And I hope you’ll be with us for our more engaging, attractive, and informative posts.

Основные функции

miniSipServer — это удобный и мощный сервер для работы с SIP/VOIP, который позволяет пользователям легко развернуть и управлять своей VoIP-системой. Он построен на открытом стандарте SIP и предлагает интуитивно понятный графический интерфейс для управления.

Совместимость и гибкость

Программа поддерживает множество популярных аппаратных и программных телефонов SIP, что избавляет от необходимости придерживаться оборудования одного производителя. miniSipServer позволяет совершать и принимать звонки через сотовые сети с использованием VoIP-шлюзов.

Бизнес-функции

С помощью miniSipServer администраторы могут предоставлять различные полезные услуги, такие как голосовая почта, группы звонков и функции «найти меня» / «следуй за мной». Это позволяет сотрудникам оставаться на связи без изменения конфигурации, независимо от их местоположения.

Управление и мониторинг

Все настройки и данные программы хранятся в каталоге установки, что упрощает управление. miniSipServer поддерживает протокол RFC3261, что позволяет использовать его для мониторинга состояния пользователей и подключения к сети оператора связи.

Дополнительные возможности

Программа включает поддержку записи детализации вызовов (CDR), системного чёрного списка и STUN. Благодаря простоте настройки, пользователи могут запустить свою VoIP-систему всего за несколько часов.

Преимущества

  • Простота использования
  • Поддержка широкого спектра SIP-устройств
  • Предоставление полезных бизнес-услуг
  • Возможность коммуникации без изменения конфигурации
  • Централизованное хранение данных
  • Мониторинг состояния пользователей
  • Быстрый запуск VoIP-системы

Скачать с официальной страницы

miniSipServer

Похожие программы

The Mizu VoIP Server Compact is a free professional softswitch for the Windows operating system with a long list of features including business modules such as pricing/billing and a convenient graphical user interface for administrators.
The server is powered by Mizutech Compact VoIP engine with an automated install and configuration wizard.

Despite its name, the Compact version doesn’t mean that this is a simplified release. Actually it contains most of the commercial version features, but using an embedded database and simplified GUI to ease the usage. Use it as a simple SIP proxy, Softswitch or as an IP-PBX depending on your needs.
Features include: SIP, routing, billing, user management, voice calls, video calls, PBX features (hold, forward, transfer, conference and many more), rich codec support (including G.729, HD audio and many more), chat, unified communication, presence, DID, SMS, voice recording and many more.

The softswitch is meant to be used also by non-technical people featuring a comprehensive documentation and intuitive user interface for all the important settings.
This software is free for non-commercial usage and it can be used with up to 20 users, 5 simultaneous calls. For direct support or commercial usage we encourage you to upgrade to one of our commercial paid license. The commercial version uses a full external SQL engine and a robust scalable core suitable for any businesses with any amount of traffic.

Sip Server

A simple sip server for handling VoIP calls based on sip protocol.

Features

Registration Of Users

The server supports registration process.
When a user send REGISTER request, the server replies with 200 OK response.
Note that the server does not validate user’s credentials.

Call Flow

The server supports a normal call flow.
When a user sends an INVITE request, the request is handled by the server and it is forward to the destination.
If the destination is busy, the server send SIP/2.0 486 Busy Here message to the source.
If the destination pick up the call, the server transfer 200 OK message to the source.
If the source want to cancel the call, the server send cancel message to the destination, and replies to the source with 200 OK message and SIP/2.0 487 Request Terminated message.
If one of the sides has been hung up, the server will got bye message which will be sending to the other side to indicate the end of the call.

Build

Windows

Using Visual Studio command prompt

  mkdir build && cd build
  cmake ..
  msbuild SipServer.sln

Linux

  mkdir build && cd build
  cmake ..
  make

Program options

--ip= The sip server ip.
--port= The sip server port. The default value is 5060.

Usage Guide

  1. Download a softphone software like Zoiper, Express Talk or any other software.
  2. Create new sip accounts and set their domain to the sip server ip.
  3. Run the sip server.
  4. Register the accounts and make calls :)

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