Upgrading VoIP platform - hardware requirements


We are considering upgrading our VoIP platform and I’m wondering what kind of hardware we need for our requirements.
The architecture will consist of a SIP Proxy Server (Kamailio) in front and a cluster of Media/Application Servers behind it.

The requirements are based on a single server.

SIP Proxy Server:

  • Up to 100 – 300 calls per second.
  • Up to 5000 – 10000 concurrent calls.
  • RTP Proxy
  • Record-Route

Media/Application Server:

  • Up to 100 – 300 calls per second.
  • Up to 5000 – 10000 concurrent calls.

What do we need in terms of CPU, Memory, HDD when looking at the following specific requirements:

  • 100 CPS
  • 200 CPS
  • 300 CPS
    Which hardware components are critical for handling this many calls, is it CPU, Memory, Disk IO bound?

Also which open source media/application server software is suitable for these requirements? We are currently using Asterisk, which works great, but perhaps there are others available which are more suitable for these requirements.

Thanks in advance!


Grant Bagdasarian

My guess is that you will exceed the OS limit on the number of open file descriptors if you try to run Asterisk with that number of concurrent calls.

I’m not otherwise knowledgeable on sizing such huge systems, but my gut feeling is that Asterisk will be limited to about an order of magnitude less.

In any case, as you are pushing the envelope, I think you will need carry out experiments with your actual use of the system.

Disk loads will depend on details of the application that you haven’t provided. Netowrk loads should be fairly easy to compute, once you know the choice of codec.

A while ago I’ve managed to get about 2000 concurrent calls on one Asterisk machine using Sipp. I had to increase the number of open file descriptors on the OS for that though. The application was simply playing an audio message which lasted about 5 seconds.
Using the following hardware:

  • Quad core Intel Xeon E5320
  • 32GB RAM

The main focus for the new VoIP platform is generating outgoing calls as quickly as possible. The applications will vary from simple audio playback to a combination of audio playback and DTMF recording.

Along with outgoing calls we will also have a small amount of incoming calls which primarily are forwarded to another remote destination.

As for the codec, G711-Alaw.