Best Self Hosted Alternatives to BroadWorks (Cisco BroadWorks)

FreeSWITCH is an open-source, software-defined telecom stack used to build and run voice, video, and real-time communications infrastructure on commodity hardware. It is commonly used as a softswitch and media server for SIP-based telephony and can also provide browser-based calling via WebRTC.

Key Features

• SIP signaling and media handling suitable for softswitch and PBX-style deployments
• Native WebRTC capabilities for browser-based real-time communications
• Modular architecture with loadable modules for telephony features and integrations
• Built-in conferencing and real-time media services (including mixing and related functions)
• Common voice features such as call routing, IVR building blocks, call recording, and voicemail support

Use Cases

• Build and operate a SIP softswitch or VoIP application server for carriers and enterprises
• Implement IVR, call routing, and contact-center style call flows
• Provide WebRTC calling to web applications without requiring a separate gateway

Limitations and Considerations

• Configuration and operation can be complex for newcomers due to the breadth of telecom concepts and module options
• Certain advanced capabilities may be available only via commercial modules or enterprise distributions

FreeSWITCH is a mature foundation for telecom and real-time media systems, from embedded devices to large-scale deployments. Its modular design and protocol support make it a flexible core for custom telephony platforms and communication products.

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.

Kamailio is an open source SIP signaling server used to build and scale VoIP and real-time communication platforms. It focuses on high performance and flexibility, and is commonly deployed as a SIP proxy, registrar, edge proxy, or signaling core in carrier and enterprise environments.

Key Features

• High-throughput SIP routing and proxying suitable for large deployments
• SIP load balancing, failover routing, and least-cost routing capabilities
• Registrar and location service for user registration and contact lookup
• Security and access control features, including authentication/authorization and TLS
• WebSocket support for WebRTC signaling
• Extensible modular architecture with integrations for multiple backend systems
• Support for IPv4 and IPv6 and various SIP transport options (UDP/TCP and more)

Use Cases

• Scaling SIP PBX, SIP-to-PSTN gateways, or media servers with a dedicated signaling layer
• Building carrier-grade SIP routing cores, including IMS/VoLTE-oriented deployments
• Deploying SIP edge proxy and SIP firewall-style protection in front of RTC infrastructure

Limitations and Considerations

• Provides SIP signaling (call control) and typically requires external media servers for RTP/media handling
• Configuration is powerful but can be complex, especially for advanced routing and multi-module setups

Kamailio is a mature SIP server with continuous development since 2001 and a large ecosystem. It is well-suited for operators and organizations that need a robust, customizable signaling component for VoIP, WebRTC, and unified communications.

Routr is a lightweight, cloud-ready SIP server that acts as a proxy, registrar, and location service for modern VoIP networks. It is designed to be programmable and scalable, making it suitable for carriers, communication service providers, and integrators.

Key Features

• Core SIP server functions: proxy, registrar, and location service
• Programmable routing with configurable strategies (intra-domain, ingress/egress, peer egress)
• Load balancing and session affinity for upstream media servers (for example Asterisk or FreeSWITCH)
• Multi-tenant and multi-domain support with domain-level access control lists
• Multiple transports: UDP, TCP, TLS, WS, and WSS
• Pluggable processors and middleware for extending routing and cross-cutting concerns
• Server management via APIs and tooling (CLI and gRPC management API)
• Flexible data sources, including Redis-backed location service and PostgreSQL

Use Cases

• Run a scalable SIP edge/proxy in front of one or more PBXs or media servers
• Build programmable SIP routing for multi-tenant VoIP platforms
• Deploy carrier-grade SIP infrastructure on Docker or Kubernetes

Routr provides a modern approach to SIP infrastructure by combining core SIP capabilities with APIs, extensibility, and cloud-native deployment patterns. It is a strong fit when you need a configurable SIP control plane that can integrate cleanly into automation and platform workflows.

Yeti-Switch is a carrier-grade VoIP softswitch focused on routing and controlling SIP traffic for telecom operators, wholesale voice providers, and enterprises. It combines signaling control, routing policy, and real-time call processing to build scalable voice platforms.

Key Features

• SIP call routing with policy controls (e.g., LCR-style routing, prefixes, routing groups)
• Multi-tenant/accounting-oriented concepts (customers/vendors, trunks, rate plans)
• Real-time call control and session handling suitable for SBC-style deployments
• Number/prefix management and routing rules for large dial-plan environments
• Call detail records (CDRs) and billing-oriented data model for rating workflows
• Operational tooling for managing gateways/trunks, dial-peers, and routing policies
• Designed for scalable deployments (separation of signaling, routing logic, and data)

Use Cases

• Wholesale voice routing platform for multiple vendors/customers
• Enterprise SIP interconnect/SBC layer to control ingress/egress traffic
• Voice termination/origination service with rating/billing data export

Limitations and Considerations

• Best suited for telecom/VoIP operators; requires SIP/telephony expertise to deploy and tune
• Billing/rating typically integrates with external processes/systems depending on operator workflow

Yeti-Switch is a good fit when you need a robust, operator-focused softswitch with strong routing primitives and telecom-centric entities (trunks, dial-peers, rates, CDRs). It is commonly used as a foundation for scalable SIP routing and interconnection networks where policy and cost-based routing are core requirements.