Difference between revisions of "Dino"

From Bicom Systems Wiki

m (Scenario 4)
m
Line 33: Line 33:
 
=Scenario 1=
 
=Scenario 1=
  
[[image:scenario1.png|image|Scenario 1 (ulaw, with and without recording]]
+
[[image:scenario1.png|image|Scenario 1 (g711, with and without recording]]
 +
 
 +
=Scenario 2=
 +
 
 +
[[image:scenario2.png|image|Scenario 2 (g711 -> g729, with and without recording]]
 +
 
 +
=Scenario 3=
 +
 
 +
[[image:scenario3.png|image|Scenario 3 (g711 -> g722, with and without recording]]
  
 
=Scenario 4=
 
=Scenario 4=
  
 
[[image:scenario4.png|image|Scenario 2 (SIP channels, with and without recording]]
 
[[image:scenario4.png|image|Scenario 2 (SIP channels, with and without recording]]

Revision as of 14:26, 25 June 2018

Introduction

This document represents Bicom Systems Lab test results for officeBOX appliance, measuring performance under various workloads.

These set of tests are done to determine how many concurrent calls can be made without calls being broken. We took 80% CPU usage as a allowed maximum, but for our requirements CPU usage should not go over 40-50% in production:

  • Scenario 1 – calls are made using g711 codec, with and without Call Recording enabled
  • Scenario 2 – calls are made with callers using g711 and were transcoded to g729, with and without Call Recording enabled
  • Scenario 3 – calls are made with callers using g711 and were transcoded to g722, with and without Call Recording enabled
  • Scenario 4 – calls are made using g711 codec and were transferred directly to queue, with and without Call Recording enabled

OfficeBOX hardware specifications:

  • AMD Embedded G series GX-412TC, 1 GHz quad Jaguar 64 bit core
  • 2GB DRAM
  • Gigabit Ethernet adapter (Intel i211AT)

Setup

For this testing we devised a method which provides us with the ability to make as many calls as we need using the configuration below.

On the OfficeBOX which is tested, we set up a trunk which is used to direct incoming calls to the extension or a queue, depending of the ongoing test.

When a call reaches extension or queue, we have the option to stay at current number of calls or to create additional calls and putting them on hold.

This method allows us to make as many calls as we want to, and it is used because it is not very convenient to use large number of phone devices for testing purposes, whether they are softphones or hardware devices. At the same time this method produces real life audio streams and signalling, emulating real life call.

Also we have placed a call to Music On Hold from an IP Phone in order to monitor the quality of audio when making all these concurrent calls. This way we have dual channel audio calls with Music on Hold playing on both sides.

Scenario 1

Scenario 1 (g711, with and without recording

Scenario 2

Scenario 2 (g711 -> g729, with and without recording

Scenario 3

Scenario 3 (g711 -> g722, with and without recording

Scenario 4

Scenario 2 (SIP channels, with and without recording