What is Freeswitch?
If you are working in VOIP / Telecom domain you might have heard and worked on software likes Asterisk, Opensips, Kamailio, etc. All these are used to develop and build VOIP services and are heavily used in the telecommunication domain. If we look at some voice/video chat applications like Skype, Zoom, WhatsApp, Viber they all use some sort of VoIP stack in the backend.
Freeswitch is a software that provides a fully configurable and customizable telecom stack for modern application
Freeswitch is open-source software that provides a production-grade telecom stack that has many applications like PBX, CallCenter Application, Sip Proxy, RTP Proxy, Audio / Video conference, etc.
As you are reading this blog Freeswitch running on cloud is serving many applications and people are using it to build complex modern telecommunication systems.
Freeswitch is so widely used because of the features and customization it offers.
Features:
- Freeswitch support VoIP, WebRTC and PSTN
- Freeswitch can run on a single core VM to a 32-core full dedicated server. It also supports a wide range of operating systems Linux, OsX, BSD, Solaris, and even windows.
- Freeswitch has full encryption support ZRTP, DTLS, SIPS, WSS
- Freeswitch has the support of a wide range of audio codecs like G722, PCMU, PCMA, Opus, Speex, AMR, GSM, VP8, VP9
- Freeswitch has a lot of inbuilt modules which can be used in multiple use cases. For example mod_conference for video and audio conferencing.
- Freeswitch is fully customizable you can use 3rd party open source modules or write your module. All modules are plug and play. It also supports a hot reload of modules.
- Freeswitch also supports SMPP (SMS).
- There is CLI support using fs_cli which gives access to a large number of commands executed on ESL.
- Freeswitch also supports automatic speech recognition and text-to-speech features using ASR and TTS modules.
Freeswitch Architecture:
Freeswitch is written in C/C++ and has a modular architecture. There is Core that provides a base system that is used by all the modules. Core doesn't have many functionalities most of the features are achieved by stand-alone modules. This modular design makes Freeswitch fully customizable.
Freeswitch modules and their brief description.
Endpoint: Telephone protocols like WebRTC, SIP/H.323, and POTS lines
Application: Performs a task such as playing audio or setting data
Application Programming Interface (API): Exports a function that takes text input and returns text output, which could be used across modules or from an external connection
Automated Speech Recognition (ASR): Interfaces with speech recognition systems
Chat: Bridges and exchanges various chat protocols
Codec: Translates between audio formats
Dialplan: Parses the call details and decides where to route the call
Directory: Connects directory information services, such as LDAP, to a common core lookup API
Event handlers: Allows external programs to control FreeSWITCH
File: Provides an interface to extract and play sound from various audio file formats
Formats: Plays audio files in various formats
Languages: Programming language interfaces used for call control
Loggers: Controls logging to the console, system log, or log files
Text-To-Speech (TTS): Interfaces with text-to-speech engines
Timers: POSIX or Linux kernel timing in applications
XML Interfaces: Uses XML for Call Detail Records (CDRs), RADIUS, CURL, LDAP, RPC, and/or SCGI
Freeswitch provides a whole lot of modules with its package we will explore them in the upcoming blogs.
Who uses Freeswitch:
Freeswitch is Extensible, Scalable, Flexible, and Stable that's why it is used by industry-leading organizations.
And a lot of organizations are also using Freeswitch to provide Communication as a Service (CAAS)
NOTE: There are many abbreviations and terms which may be unknown to someone who is new to the telecommunication domain, so don’t get confused a simple google search for these terms will clear their meaning.
Links:
Freeswitch docs: https://freeswitch.org/confluence/
Freeswitch Github: https://github.com/signalwire/freeswitch/
