
Asterisk
Open-source telephony engine and PBX framework

Asterisk is an open-source telephony engine and PBX toolkit implemented primarily in C and developed for GNU/Linux. It exposes traditional PBX features and low-level telephony primitives so developers and operators can build SIP, WebRTC and PSTN-connected communications applications.
Key Features
- Modular, channel-based architecture with pluggable modules for SIP (chan_pjsip / chan_sip), media, codecs and hardware interfaces
- WebRTC support (WSS/DTLS-SRTP), RTP/RTCP handling, and modern codec support including Opus for browser and realtime audio
- ARI (Asterisk REST Interface) exposing REST + WebSocket events for building custom programmable call applications
- AMI and AGI interfaces for management, automation and traditional dialplan scripting; full CLI and menuselect build configuration
- PSTN and telephony hardware integration (traditional telephony cards and drivers) alongside VoIP gateway capability
- Source-driven build system using autoconf/Autotools and GNU Make; extensive documentation, community forum and release advisories
Use Cases
- Deploying an enterprise or branch PBX providing calls, voicemail, conferencing, queues and call routing
- Acting as a VoIP gateway or SBC to bridge SIP/WebRTC clients with PSTN trunks and telephony hardware
- Building programmable communications services (IVR, voicebots, conferencing, call recording) using ARI or AMI
Limitations and Considerations
- Nontrivial operational complexity: requires careful configuration, dependency management and familiarity with telephony concepts
- Requires proactive security and performance tuning (file-descriptor limits, TLS/DTLS configuration); security advisories are periodically published for critical fixes
- Feature surface is large and modularity means some functionality requires enabling/building specific modules or external libraries
Asterisk is a mature, widely adopted telephony engine suited for operators and developers who need deep control over call handling and media. It is maintained by a large community and is intended for production PBX and programmable-telephony deployments.
Categories:
Tags:
Tech Stack:
Similar Services

Mumble
Low-latency, high-quality VoIP voice chat client and server
Mumble is an open-source, low-latency VoIP voice chat platform with a desktop client and a self-hosted server for encrypted group communication.

ejabberd
Scalable real-time messaging server for XMPP, MQTT, and SIP
ejabberd is an Erlang/OTP-based messaging server providing XMPP chat and presence, MQTT broker capabilities for IoT, and SIP services for real-time communications.
FreeSWITCH
Software-defined telecom stack for VoIP, SIP, and real-time media
FreeSWITCH is an open-source software-defined telecom stack for building VoIP and real-time communication services with SIP and WebRTC support.


Kamailio
High-performance SIP server for VoIP and real-time communications
Kamailio is an open source SIP signaling server for scalable VoIP and real-time communication platforms, supporting routing, load balancing, WebRTC, and IMS/VoLTE.
HOMER (SIPCAPTURE)
Packet capture and observability for SIP, VoIP, and WebRTC
Carrier-grade SIP/VoIP/WebRTC packet capture and monitoring platform with real-time search, correlation, and troubleshooting workflows using the HEP/EEP protocol.
Routr
Programmable SIP proxy, registrar, and location server
Routr is a programmable SIP server (proxy/registrar/location) for building scalable VoIP infrastructure with APIs, multi-tenancy, and Kubernetes-ready deployment.
Autotools
GNU Make
WebRTC
C
Linux