SIP Trunking vs SIP Proxy: When to Use Kamailio vs OpenSIPS for Your VoIP Stack

When it comes to the most prevalent technical discussion within VoIP design, the question will always arise on whether a SIP proxy or SIP trunk is needed in the VoIP architecture. In case the former option seems better, the next step would be to consider Kamailio or OpenSIPS as a platform to implement it.

The importance of making the right decision during the design phase is underestimated by many users, while later problems may arise. Inefficient work or missing certain features and costly refactoring will occur in case of choosing the wrong way.

SIP Trunking vs SIP Proxy: Understanding the Fundamental Difference

While these phrases are synonymous in everyday conversation, they mean very different things within a VoIP stack.

What is SIP Trunking?

SIP Trunking Services is the term used to denote the interconnect of the enterprise’s PBX/UC systems with the PSTN through an Internet Protocol link. In essence, a SIP trunk represents a virtual voice channel, which can be transmitted either over the Internet or a dedicated IP line.

When a company replaces its PRI or analog voice lines with SIP trunks, it means that its PBX is being connected to SIP Trunk Provider’s services. It allows them to relay traffic both to and from PSTN on behalf of the organization.

SIP Trunking key characteristics:

• Serves as the interface of your proprietary telecom infrastructure with the outside world
• Supplied by the carrier or SIP Trunk provider as a service
• Serviced in accordance with the per-channel or per-minute billing model
• Provides PSTN origination and termination capabilities
• Configured on the carrier/provider level, not the infrastructure one

What Is a SIP Proxy Server?

A SIP proxy server is part of your VoIP network structure that passes SIP traffic between the endpoints. This element will not allow access to the PSTN from your VoIP infrastructure. The SIP proxy will route SIP traffic from the endpoints, PBXs, media servers, and other components of your VoIP structure.

Key characteristics of a SIP proxy:

• Located in the VoIP network structure
• Routes requests for SIP invite, register, etc.
• Capable of performing additional operations such as load balancing, failover, authentication, and routing
• Does not offer PSTN connection
• Installed on your network devices

In reality, most VoIP solutions incorporate both SIP proxy and SIP trunking in their configuration.

Where Kamailio and OpenSIPS Fit In

Kamailio and OpenSIPS are both open-source proxy servers for SIP. They do not operate against any SIP trunk providers. Instead, these are simply the applications that make up your SIP routing layer.

Both Kamailio and OpenSIPS can trace their family history back to the same roots. OpenSIPS is a fork of SER, as is Kamailio, formerly called OpenSER. Since 2008, they have grown independently, and each is now its own distinct platform with its own philosophy and capabilities.

Kamailio: Architecture and Strengths

The main feature of our Kamailio development services is a platform that is focused on performance, reliability, and lightweight resource consumption. Our product, Kamailio, is extensively used by carriers and wholesale providers of high-volume VoIP services.

Core Strengths of Kamailio

Raw SIP processing performance: Kamailio is repeatedly rated as the most high-performing SIP proxy server. It processes millions of SIP requests per second and is a primary choice for high-volume carrier routing.

Memory efficiency: Kamailio architecture is engineered to process the maximum number of transactions with minimum memory overhead. The thing becomes especially important for running thousands of simultaneous sessions.

Stability at scale: Kamailio has been working in a carrier environment for more than two decades now. Its codebase was tested to work in real-life carrier settings and became very stable as a result.

Modular configuration: Kamailio uses its own scripting language for configuration. Experienced developers can fine-tune any part of SIP transaction processing.

Strong DIAMETER and SS7 integration: Kamailio allows bridging between VoIP and conventional telecommunications environments through support of these protocols.

Typical Kamailio Use Cases

• Class 4 softswitch deployments (wholesale carrier routing)
• Large-scale SIP trunking platform backends
• Carrier interconnect and peering points
• High-volume registration servers
• IMS (IP Multimedia Subsystem) deployments
• ENUM routing

OpenSIPS: Architecture and Strengths

OpenSIPS Development Services is based on the platform, which is focused on flexibility, richness in functionality and capability to deploy advanced application logic through the SIP proxy layer. OpenSIPS’ development trend is oriented towards full-fledged application servers, whereas Kamailio remains true to its proxy nature.

Key Strengths of OpenSIPS

Advanced call logic and scripting: The most important characteristic of OpenSIPS is that it has richer possibilities to develop scripts and perform complex call processing logic right on the SIP proxy level.

Built-in presence and IM support: In addition, presence, instant messaging and advanced communication features are provided out-of-the-box by OpenSIPS. Hence, it is perfectly suited for UC deployment.

REST and MI interfaces: OpenSIPS has a powerful built-in management interface (MI) and RESTful API, allowing it to integrate applications and perform billing and monitoring tasks easily.

Topology hiding and B2BUA: OpenSIPS is well known for its B2BUA capabilities. They allow implementing sophisticated routeing scenarios when you need to route two independent legs of a call.

Active-active clustering: Finally, OpenSIPS provides a very convenient clustering mechanism, which allows deploying HA architectures without using any database layer.

Typical OpenSIPS Use Cases

• Class 5 softswitch (retail subscriber routing with complex features)
• UCaaS and hosted PBX platforms
• Contact center routing engines
• WebRTC gateways
• Presence and instant messaging servers
• SIP trunking platforms with complex routing logic

Kamailio vs OpenSIPS: A Direct Comparison

Decision Framework: Which Platform Is Right for Your Project?

Choose Kamailio When:

The application in development requires handling millions of SIP messages per second, with very low latency. Kamailio is regarded as the industry standard for this type of solution because of its performance capabilities.

The application requires DIAMETER or legacy telecommunication protocols. Kamailio has better support modules than any other technology available for legacy telecom signaling.

Your company has developers familiar with Kamailio, or you are working with a partner that offers Kamailio development services. More complex to configure but much more powerful than other technologies for pure proxy use cases.

The application being developed is a class 4 softswitch, where the main task will be the optimal routing of calls.

Choose OpenSIPS When:

Your deployment requires that call features, presence capabilities, and routing logic be of equal importance to the overall call handling capacity of the system.

You require close integration of your system with other applications using RESTful APIs. The OpenSIPS management console will enable this to be simpler.

Your application includes a WebRTC/real-time communication application in which you require B2BUA capabilities of call legs.

Your project is to build a retail subscriber platform (Class 5), in which personal user features, routing of voicemails, routing of call forwardings, and presence control are major considerations.

Your developers find the OpenSIPS scripting capability to suit your needs.

SIP Proxy Server in a Complete VoIP Stack

Kamailio and OpenSIPS do not operate independently. Typically, a deployed VoIP architecture consists of several building blocks:

SIP Proxy Server (Kamailio or OpenSIPS): SIP proxying, load balancing, authentication, and application logic are performed here.

Media Server (FreeSWITCH or Asterisk): The actual audio work is done by this module, including transcoding, conferencing, and interactive voice response (IVR).

SBC (Session Border Controller): Serves as the network boundary, dealing with security, NAT traversal, and interconnect.

Billing and Rating Engine: Responsible for collecting and managing call detail records.

Database Layer: This layer stores subscriber and call data.

These modules have their specific purposes. A SIP proxy server works with signaling. The media server takes care of audio. SBC handles the boundaries. Mixing up these responsibilities results in poorly designed architectures.

Combining SIP Trunking with Your Proxy Layer

After installing the internal routing layer within your system, the next step involves connecting it to the PSTN using SIP trunking solutions. The SIP proxy will take care of internal route management and direct your outgoing calls to the SIP trunk provider. Incoming calls from the SIP trunk provider get directed to your proxy, which routes them appropriately.

Trunk failover management can be easily managed using your SIP proxy. Whenever your primary SIP trunk provider suffers from a failure, then your proxy will automatically route your call traffic to the secondary trunk provider without involving any form of manual work. The failover mechanism is among the best you can implement in Kamailio/OpenSIPS.

Performance Considerations

In the case of large-scale deployments, some key design considerations will strongly influence the performance no matter what platform you go for:

Stateless or stateful operation: While Kamailio and OpenSIPS support both stateless (SIP forwarding only) and stateful (transaction state tracking) modes of operation, the former is clearly faster but leaves much less flexibility when it comes to writing your routing logic.

Database interactions: Database access involves extra latency. Reduce the number of synchronous database accesses in your SIP processing route via using some kind of caching mechanism (in-memory htable module in Kamailio and cacheDB module in OpenSIPS).

Configuration of the load balancer: Both Kamailio and OpenSIPS support distributed load balancing schemes. Try to distribute the load in such a way as to avoid bottlenecks at any one node.

Choice of hardware: In the context of large-scale SIP routing, CPU clock speed and single-core performance become more important than the number of cores. The choice of network adapter hardware is extremely important. Tuning the OS can increase throughput two times compared to a vanilla configuration.

Frequently Asked Questions

Q: Can we use both Kamailio and OpenSIPS in the same stack?

A: Yes, some architectures use Kamailio at the carrier edge for high-performance SIP routing and OpenSIPS closer to the application layer where more complex call logic is needed. This is not uncommon in large-scale deployments.

Q: Which platform has better community support?

A: Both have active communities and regular release cycles. Kamailio has a larger installed base in carrier environments. OpenSIPS tends to have stronger representation in UCaaS and CPaaS developer communities.

Q: Do I need a SIP proxy if I am using a commercial SIP trunk service?

A: Depends on your scale and requirements. A small business using a single SIP trunk for a handful of users may not need a proxy. Any deployment with multiple trunks, complex routing, failover requirements, or more than a few dozen users benefits from a dedicated SIP proxy.

Q: How does licensing work for Kamailio and OpenSIPS?

A: Both are open-source under GPL licenses. You can use, modify, and deploy them freely. Commercial support contracts are available from various vendors, including Dialiqo.

Q: What databases do these platforms work with?

A: Both support MySQL, PostgreSQL, Redis, Cassandra, MongoDB, and others through their module systems. The right database choice depends on your performance requirements and existing infrastructure.

Conclusion

It should be understood that SIP Trunking Solutions and the SIP Proxy Server do not compete against each other because they address different aspects of your VoIP infrastructure. The question that you need to address is whether Kamailio Development Services or OpenSIPS Development Services are better suited to meet your needs at the proxy server level based on your requirements and traffic pattern.

It can be safely concluded that carriers and heavy-duty routing systems typically prefer Kamailio, while UCaaS services, contact centers, and feature-laden hosting solutions prefer OpenSIPS. Of course, one system cannot be considered inherently better than the other.

Build Your VoIP Routing Layer with Dialiqo

At Dialiqo, we have professionals with practical experience with Kamailio development services as well as OpenSIPS development services. Our professionals assist our clients in building SIP proxy server solutions tailored to their needs in terms of capacity, functionality, and cost.

No matter what SIP proxy architecture solution you require (e.g., carrier solution, hosted PBX solution, contact center routing solution), we will assist you in selecting appropriate solutions and implementing the proper infrastructure.

Contact Dialiqo to get help with VoIP proxy server solutions.

Check dialiqo.com to learn more about our services regarding SIP proxies.