How to Choose Between FreeSWITCH, Asterisk, Kamailio, and OpenSIPS for Your Project

When developing your own VoIP application from the ground up, or when reviewing an existing solution stack, there is no doubt that you are going to come across all four open-source platforms mentioned above: FreeSWITCH, Asterisk, Kamailio, and OpenSIPS. These solutions make up the backbone of open source VoIP development and have substantial user bases and communities.

However, this doesn’t mean that all four options are interchangeable. In fact, each solution is built with certain capabilities in mind and performs best when used according to its design. Selecting the wrong platform for your needs would mean going against the platform rather than using its inherent abilities. This article will show the strengths and weaknesses of each VoIP software.

Understanding the Two Categories

There is a vital point to make before proceeding to compare specific tools.

Media servers, including FreeSWITCH and Asterisk, are responsible for the actual audio layer. They terminate SIP calls, transcode audio between different codecs, manage conferencing, perform IVR processing and recording, and provide media content to the callers. In simple words, they process the voice itself.

SIP proxy servers such as Kamailio and OpenSIPS are responsible for signaling SIP messages from one endpoint to another, managing user registration and logins, applying complex logic for routing messages, performing load balancing, and authenticating users.

VoIP stacks used in real-life production systems often employ both media and signaling layers, where the former processes calls while the latter handles signaling. It is essential to know the difference between these concepts because otherwise, you can easily make a crucial mistake.

FreeSWITCH: The Modern Media Server

FreeSWITCH development revolves around a platform explicitly created as an architecture for a scalable, multi-tenant media server. FreeSWITCH was launched in the year 2006 by a team of Asterisk veterans looking for a simpler architecture, and FreeSWITCH operates using a multithreaded framework.

What FreeSWITCH Does Well

• High-concurrency call handling on a single node (thousands of simultaneous calls)
• Native WebRTC support with built-in DTLS-SRTP handling
• Multi-tenant architecture with per-domain configuration isolation
• Powerful ESL (Event Socket Layer) for programmatic call control via any language
• Built-in conference server with extensive feature support
• Excellent codec support and transcoding capability
• XML-based configuration that scales well for complex deployments
• Strong support for video calls and video conferencing

When to Choose FreeSWITCH

• You are building a hosted PBX or UCaaS platform with multiple tenants
• WebRTC is a core requirement
• You need programmatic call control via API (chatbots, voicebots, IVR)
• Your project requires video calling or conferencing features
• You need a media server that can scale vertically to high call concurrency

FreeSWITCH Limitations

Configuring FreeSWITCH using XML may be cumbersome for complex environments. However, it is not ideal for use as a SIP proxy in high-volume traffic that doesn’t have media involved. Another aspect that is quite different is the multi-threading of FreeSWITCH compared to Asterisk.

Asterisk: The Established Standard

Asterisk Development focuses on the product that created the open-source telephony market. Introduced in 1999, Asterisk is the most widely deployed open-source VoIP solution. More VoIP architects have experience with the Asterisk dial plan than any other system for configuring call routing rules.

What Asterisk Does Well

• Extensive module ecosystem covering almost every telephony feature imaginable
• Familiar dial plan syntax that most VoIP engineers already know
• Strong PSTN integration via TDM hardware and SIP trunks
• ARI (Asterisk REST Interface) for modern application development
• AGI scripting for tight integration with business logic
• Mature voicemail, call queuing, and IVR capabilities
• Large community and extensive documentation

When to Choose Asterisk

• Your team has existing Asterisk expertise or your developers come from a traditional telephony background
• You need broad hardware compatibility, including TDM cards for PSTN connectivity
• Your project is a business phone system, call center, or PBX replacement
• You need extensive module support without custom development
• Your call volumes are moderate and you can scale horizontally

Asterisk Limitations

Due to its single-threaded approach to handling channels, an aspect that was modified in newer versions, vertical scalability is a weak point compared to FreeSWITCH. Asterisk installations with high concurrency require horizontal scalability, using several machines behind a proxy setup. WebRTC implementation has advanced significantly, but still trails FreeSWITCH’s.

Kamailio: The Carrier-Grade SIP Proxy

Kamailio Development focuses on an underlying platform designed specifically for high-speed SIP routing. Kamailio (formerly OpenSER) is the industry-standard SIP proxy for wholesale carriers and large SIP trunking environments where millions of SIP transactions have to be routed every second.

What Kamailio Does Well

• Industry-leading SIP routing throughput
• Memory-efficient handling of massive registration tables
• Script-based configuration with fine-grained control over every SIP message
• Mature load balancing and failover across media server farms
• Strong DIAMETER and SS7 integration for traditional telecom environments
• Proven stability in carrier-grade deployments over decades

When to Choose Kamailio

• You are building a wholesale SIP routing platform or Class 4 softswitch
• Your primary concern is SIP message throughput and routing efficiency
• You need a front-end proxy for a farm of FreeSWITCH or Asterisk nodes
• Your team includes engineers familiar with Kamailio scripting
• You need telecom-grade stability and carrier interconnect capabilities

OpenSIPS: The Feature-Rich SIP Proxy

The OpenSIPS Development Project relies on a framework that was derived from the same sources as Kamailio yet shifted to an application server architecture. The main areas of use for OpenSIPS include SIP retail services, UCaaS solutions, and contact center routing with complicated call-handling capabilities.

What OpenSIPS Does Well

• More expressive scripting environment for complex call logic
• Built-in B2BUA for direct control of both call legs
• Strong presence and instant messaging support
• Well-developed REST/MI management interface
• Native active-active clustering for high availability
• Good fit for WebRTC-to-SIP gateway architectures

When to Choose OpenSIPS

• You are building a hosted PBX backend or UCaaS routing layer
• Complex routing logic, presence, and subscriber features are requirements
• You need a B2BUA without deploying a full media server
• Your team finds OpenSIPS scripting more accessible for your use cases
• You are building a contact center routing platform with dynamic agent logic

Comparison Matrix

Combining Platforms: The Real-World Architecture

Production VoIP stacks rarely use just one platform. Common combinations include:

Kamailio + FreeSWITCH

Kamailio handles SIP signaling, load balancing, and registration at scale. FreeSWITCH handles media for calls that need processing. This is a standard architecture for hosted PBX and UCaaS platforms.

OpenSIPS + Asterisk

OpenSIPS handles routing logic and subscriber management. Asterisk handles specific call flows that require complex dial plan logic or hardware integration. Common in mid-market business telephony.

Kamailio + OpenSIPS

Used in very large deployments where Kamailio handles the high-volume edge routing and OpenSIPS handles application-layer logic closer to the subscriber tier.

Frequently Asked Questions

Q: Can one developer work on all four platforms?

Many VoIP engineers have working knowledge of all four, but deep expertise in any one of them takes significant time. For a new project, it is more efficient to choose the right platform for the job and find developers with that specific expertise than to force a familiar tool into an unsuitable role.

Q: Which platform is easiest to start with?

Asterisk has the lowest learning curve for someone with a traditional telephony background. FreeSWITCH has a steeper initial curve but a more flexible architecture for complex applications. Kamailio and OpenSIPS both require solid SIP knowledge before the configuration makes intuitive sense.

Q: How do we decide between FreeSWITCH and Asterisk specifically?

If WebRTC, multi-tenancy, and high call concurrency on a single node are priorities, FreeSWITCH is the better fit. If your team has existing Asterisk expertise, your use case is a traditional PBX or call center, and PSTN hardware integration matters, Asterisk is the better fit.

Q: Is there a commercial support option for these platforms?

Yes. All four platforms have commercial support options available from various vendors. Dialiqo provides development and support services for all four platforms.

Q: What about licensing?

FreeSWITCH, Kamailio, and OpenSIPS are licensed under MPL or GPL. Asterisk is licensed under GPL with a commercial license option (Asterisk Business Edition, from Sangoma). All can be deployed commercially, but review the specific license terms for your use case.

Get Expert Help Choosing and Building Your VoIP Stack

Dialiqo boasts extensive experience in FreeSWITCH, Asterisk, Kamailio, and OpenSIPS software. We assist organizations in selecting the most suitable technology and deliver robust VoIP systems according to their demands.

Get in touch with Dialiqo to explore your VoIP infrastructure needs. For more information on our VoIP development solutions, visit dialiqo.com.