Real Time Communication Tools

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Real-time communications (RTC) is any mode of telecommunications in which all users can exchange information instantly or with negligible latency or transmission delays. In this context, the term real-time is synonymous with life. RTC always has a direct path between the source and the destination. Real-time communication is a category of software protocols and communication hardware media that gives real-time guarantees, which is necessary to support real-time computing guarantees. It enables employees from across the world to communicate with each other 24×7 and share ideas or solve problems quickly. It is a cost-effective way of getting several people from different locations to attend meetings and conferences – without spending time or money on travel and accommodation.


FreeSWITCH is a free, open-source application server for real-time communication, WebRTC, telecommunications, video, and Voice over Internet Protocol. It is a multiplatform server that runs on Linux, Windows, macOS, and FreeBSD. FreeSWITCH is used to build and develop PBX systems, IVR services, videoconferencing platforms, collaboration stations that include chat and screen sharing, and Session Border Controllers (SBCs) and voice gateways. It has full support for encryption, ZRTP, DTLS, and SIPS and can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.


Asterisk turns an ordinary computer into a communications server: Asterisk powers IP PBX systems, VoIP gateways, conference servers, and other custom solutions. Small businesses use it, large businesses, call centers, carriers, and government agencies, worldwide. Asterisk is free and open source. You can extend your existing phone system or use it as a complete solution. The Asterisk PBX utilizes SIP Trunking, allowing you to have a fully customized communication application. If you need a communication platform that can accommodate a changing system, Asterisk will fulfill your requirements.


Kamailio (successor of former OpenSER and SER) is an Open Source SIP Server released under GPLv2+, handling thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and real-time communications – presence, WebRTC, Instant messaging, and other applications. Moreover, Kamailio can easily use it for scaling up SIP-to-PSTN gateways, PBX systems, or media servers like Asterisk, FreeSWITCH, or SEMS.

Among the powerful features: asynchronous TCP, UDP, and SCTP; secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication, and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached; Json and XMLRPC control interface, SNMP monitoring.


OpenSIPS is an Open Source SIP proxy/server for voice, video, IM, presence, and other SIP extensions. OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms, or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others.


Routr is a Next-generation SIP Server and Lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. Routr ships with all the tools necessary to deploy your VoIP network. A command line is included to control the server remotely. You can also control the server using the RESTful API or the GUI from the comfort of the web browser.

Sippy B2BUA

Sippy B2BUA is an RFC3261-compliant Session Initiation Protocol (SIP) Back-to-back user agent (B2BUA). The Sippy B2BUA is a SIP call controlling component. Unlike a SIP proxy server, which only maintains a transaction state, the Sippy B2BUA maintains a complete call state and participates in all call requests. The Sippy B2BUA is a SIP call controlling component. Unlike a SIP proxy server, which only maintains a transaction state, the Sippy B2BUA maintains a complete call state and participates in all call requests. For this reason, it can perform several functions that are impossible to implement using a SIP proxy, such as accurate call accounting, pre-paid rating and billing, fail-over call routing, etc. Unlike PBX-type solutions such as Asterisk, for example, the Sippy B2BUA doesn’t perform any media relaying or processing. Therefore it doesn’t introduce any additional packet loss, delay, or jitter into the media path.


Flexisip is a complete, modular, and scalable SIP server suite written in C++11, comprising proxy, presence, and group chat functions.

Flexisip offers an easy-to-install SIP server solution, offering all the features required to deploy your SIP service tuned for mobile or desktop applications “out of the box.”

Administrators can deploy flexisip instances on servers to run a SIP VoIP service.
The free SIP service has run on Flexisip since 2011 and enables Linphone users to create SIP addresses to connect.


Janus is a WebRTC Server developed by Meetecho and conceived as a general-purpose one. As such, it doesn’t provide any functionality per se other than implementing the means to set up a WebRTC media communication with a browser, exchanging JSON messages with it, and relaying RTP/RTCP and messages between browsers and the server-side application logic they’re attached to. Any specific feature/application is provided by server-side plugins that browsers can contact via Janus to take advantage of their functionality. Examples of such plugins can be implementations of applications like echo tests, conference bridges, media recorders, SIP gateways, and the like.


Maxim Sobolyev initially developed RTPProxy in 2003 to enable VoIP calls to/from SIP User Agents located behind NAT firewalls. In several cases, direct end-to-end communication between NATed SIP User Agents was impossible. RTPProxy, in conjunction with a SIP proxy, overcomes these impediments by acting as a go-between for the RTP streams.

Subsequently, the RTPProxy became widely used by VoIP service providers that needed to optimize network traffic flow.

Later, it became apparent that this software has many other possible uses. It can be used with a signaling element (SIP Proxy or SIP B2BUA) to build complex VoIP networks, optimize traffic flow, collect voice quality information, and so on. It can perform many additional functions on RTP streams, including call recording, playing pre-encoded announcements, real-time stream copying, and RTP payload re-framing.

The RTPproxy supports advanced features, such as remote control mode, allowing for scalable distributed SIP VoIP networks to be built. The nathelper module included in the SIP Express Router (SER), OpenSIPS or Kamailio, and Sippy B2BUA allow using multiple RTPproxy instances running on remote machines for fault-tolerance and load-balancing purposes.


The Sipwise NGCP rtpengine is a proxy for RTP traffic and other UDP-based media traffic. It’s meant to be used with the Kamailio SIP proxy and forms a drop-in replacement for any other available RTP and media proxies.

Currently, the only supported platform is GNU/Linux.


An SFU (Selective Forwarding Unit) receives audio and video streams from endpoints and relays them to everyone else (endpoints send one and receive many). Each receiver endpoint can select which streams and spatial/temporal layers it receives. Compared to a mixer or MCU (Multipoint Conferencing Unit), this design leads to better performance, higher throughput, and less latency. It’s highly scalable and requires much fewer resources, given that it does not transcode or mix media. Since endpoints get the other endpoints’ media separately, they can have a personalized layout and choose which streams to render and how to display them.


SEMS is a free, high-performance, extensible media server for SIP (RFC3261) based VoIP services. It is intended to complement proxy/registrar servers in VoIP networks for all applications where server-side audio processing is required, for example, away or pre-call announcements, voicemail, or network-side conferencing. Another use case is for interconnecting SIP networks, where a back-to-back user agent (B2BUA) is required.


Jitsi is a collection of free and open-source multiplatform voice, video conferencing, and instant messaging applications for the web platform Windows, Linux, macOS, iOS, and Android. The Jitsi project began with the Jitsi Desktop.


Microsoft Teams is a proprietary business communication platform developed by Microsoft as part of the Microsoft 365 family of products. Teams primarily compete with the similar service Slack, offering workspace chat and videoconferencing, file storage, and application integration.


Slack is a messaging app for businesses that connects people to the information they need. Slack transforms organizations’ communication by bringing people together to work as a unified team. Slack makes access to your colleagues easy — message anyone inside or outside your organization and collaborate just like you would in person. People can work in dedicated spaces called channels that bring together the right people and information. Slack supports asynchronous work. When work is organized in channels, no matter your location, time zone, or function, you can access the information you need on your own time. Ask questions, get caught up, and share updates without having to coordinate schedules.


Webex by Cisco is an American company that develops and sells web conferencing, videoconferencing, unified communications as a service, and contact center as a service application. It was founded as WebEx in 1995 and taken over by Cisco Systems in 2007. Its headquarters are in San Jose, California.

Vonage Meetings

A reliable and more secure video conferencing solution built into Vonage Business Communications (VBC) that you can trust. Enjoy virtual group settings and keep large audiences engaged face-to-face. The Speaker view brings the active speaker to the forefront of the discussion. Participants can seamlessly access screen share content and files with a single click. Video thumbnails showcase all participants in a customizable view. Video-first experience that automatically adapts to meet the size of your meeting. Keep everyone engaged using Roundtable and Watch Together.


Zoom is a communications platform that allows users to connect via video, audio, phone, and chat. Using Zoom requires an internet connection and a supported device. Zoom is one of the leading video conferencing software apps. It enables you to interact virtually with co-workers when in-person meetings aren’t possible, and it has also been hugely successful for social events.


Discord is a VoIP and instant messaging social platform. Users can communicate with voice calls, video calls, text messaging, media, and files in private chats or as part of communities called servers.


Element is a free and open-source software instant messaging client implementing the Matrix protocol. Element supports end-to-end encryption, groups, and sharing of files between users. It is available as a web application, as a desktop app for all major operating systems, and as a mobile app for Android and iOS.

Blue Jeans

BlueJeans by Verizon is a company that provides an interoperable cloud-based video conferencing service. It is headquartered in the Santana Row district of San Jose, California. Before being acquired by Verizon, the company was known as BlueJeans Network.

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