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Asterisk Kamailio Voice over IP

Kamailio vs Asterisk

Comparison on Asterisk and Kamailio and where they fit in the Telephony landscape.

One of the most searched keywords that leads to this site is Kamailio vs Asterisk, so I thought I’d expand upon this a bit more as I’m a big fan of both, and it’s somewhat confusing.

(Almost everything in this post I talk about on Asterisk is roughly true for FreeSWITCH as well, although FS is generally more stable and scalable than Asterisk. )

Asterisk

Asterisk is a collection of PBX / softswitch components that you can configure and put together to create a large number of different products with the use of config files and modules.

Asterisk can read and write the RTP media stream, allowing it to offer services like Voicemail, B2B-UA, Conferencing, Playing back audio, call recording, etc.

It’s easy to learn and clear to understand how it handles “calls”.

Kamailio

Kamailio is a SIP proxy, from which you can modify SIP headers and then forward them on or process them and generate a response.

Kamailio is unable to do manipulate the RTP media stream. It can’t listen to, modify or add to the call audio, it only cares about SIP and not the media stream. This means it can’t playback an audio file, record a call or serve voicemail.

Kamailio has a bit of a steep learning curve, which I’ve tried to cover in my Kamailio 101 series, but even so, Kamailio doesn’t understand the concept of a “call”, it deals in Sessions, as in SIP, and everything you want to do, you have to write into Kamailio’s logic. Awesome power but a lot to take in.

Note – RTPengine is growing in capabilities and integrates beautifully into Kamailio, so for some applications you may be able to use RTPengine for media handling.

ScaleSpeedStabilityMedia FunctionsEase
AsteriskXX
KamailioXXX

Working Together

Asterisk has always had issues at scale. This is for a variety of reasons, but the most simplistic explanation is that Asterisk is fairly hefty software, and that each subscriber you add to the system consumes resources at a rate where once your system reaches a few hundred users you start to see issues with stability.

Kamailio works amazingly at scale, it’s architecture was designed with running at scale in mind, and it’s super lightweight footprint means the load on the box between handling 1,000 sessions and handling 100,000 sessions isn’t that much.

Because Asterisk has the feature set, and Kamailio has the scalability, so the the two can be used together really effectively. Let’s look at some examples of Asterisk and Kamailio working together:

Asterisk Clustering

You have a cluster of Asterisk based Voicemail servers, serving your softswitch environment. You can use a Kamailio instance to sit in front of them and route INVITEs evenly throughout the cluster of Asterisk instances.

You’d be using Asterisk’s VM functions (because Asterisk can do media functions) and Kamailio’s SIP routing functions.

Here’s an example of Kamailio Dispatcher acting in this function.

Application Server for SIP Softswitch

You have a Kamailio based Softswitch that routes SIP traffic from customers to carriers, customers want a hosted Conference Bridge. You offer this by routing any SIP INVITES to the address of the conference bridge to an Asterisk server that serves as the conference bridge.

You’d be using Kamialio to route the SIP traffic and using Asterisk’s ability to be aware of the media stream and join several sources to offer the conference bridge.

Which should I use?

It all depends on what you need to do.

If you need to do anything with the audio stream you probably need to use something like Asterisk, FreeSwitch, YaTE, etc, as Kamailio can’t do anything with the audio stream.*

If it’s just signalling, both would generally be able to work, Asterisk would be easier to setup but Kamailio would be more scaleable / stable.

Asterisk is amazingly quick and versatile when it comes to solving problems, I can whip something together with Asterisk that’ll fix an immediate need in a faction of the time I can do the same thing in Kamailio.

On the other hand I can fix a problem with Kamailio that’ll scale to hundreds of thousands of users without an issue, and be lightning fast and rock solid.

Summary

Kamailio only deals with SIP signalling. It’s very fast, very solid, but if you need to do anything with the media stream like mixing, muxing or transcoding (RTP / audio) itself, Kamailio can’t help you.*

Asterisk is able to deal with the media stream, and offer a variety of services through it’s rich module ecosystem, but the trade-off is less stability and more resource intensive.

If you do require Asterisk functionality it’s worth looking into FreeSWITCH, although slightly harder to learn it’s generally regarded as superior in a lot of ways to Asterisk.

I don’t write much about Asterisk these days – the rest of the internet has that pretty well covered, but I regularly post about Kamailio and other facets of SIP.

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