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='''Introduction'''=
 
  
This document represents Bicom Systems Lab results for [[Appliances#OfficeBOX| officeBOX]] appliance, measuring call performance under various workloads.
 
 
These set of tests are done to determine how many concurrent calls can be made without audio being broken. We set 80% CPU usage as a technically allowed CPU usage maximum, but for our requirements and safe usage of the appliance, we limit the number of calls to 60% of the maximum number of calls that previous technical limitation allows. This included four scenarios in testing:
 
 
*[[#Scenario_1: G711| Scenario 1: G711]] – calls are made using g711 codec, with and without Call Recording enabled
 
 
*[[#Scenario_2| Scenario 2: G711->G729]] –  calls are made with callers using g711 and were transcoded to g729, with and without Call Recording enabled
 
 
*[[#Scenario_3| Scenario 3: G711->G722]] – calls are made with callers using g711 and were transcoded to g722, with and without Call Recording enabled
 
 
*[[#Scenario_4| Scenario 4: Queue channels]] – 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 following configuration.
 
 
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 on the ongoing test.
 
 
When a call reaches an extension or a queue, we have the option to read usage statistics at current number of calls or to create additional calls for further straining of the device.
 
 
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 audio streams and signalling, simulating real life calls.
 
 
For audio quality, we have placed a call to Music On Hold from an IP Phone in order to monitor the 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: G711'''=
 
 
In scenario 1 calls are made using g711 codec, and we made two tests with and without Call Recording enabled. This gives the following results:
 
 
[[image:scenario1.png|image|Scenario 1 (g711, with and without recording]]
 
 
='''Scenario 2'''=
 
 
Tests in scenario 2 feature calls that originate with g711 codec and were transcoded to g729, with and without Call Recording enabled, which gives following results:
 
 
[[image:scenario2.png|image|Scenario 2 (g711 -> g729, with and without recording]]
 
 
='''Scenario 3'''=
 
 
Scenario 3 – calls are made with callers using g711 and were transcoded to g722, with and without Call Recording enabled
 
 
[[image:scenario3.png|image|Scenario 3 (g711 -> g722, with and without recording]]
 
 
='''Scenario 4'''=
 
 
Scenario 4 – calls are made using g711 codec and were transferred directly to queue, with and without Call Recording enabled
 
 
[[image:scenario4.png|image|Scenario 2 (SIP channels, with and without recording]]
 

Latest revision as of 16:30, 16 December 2019