3.4.1 Definitions
In the present paragraph we want to deal with Network checks, which happens to be a wide subject. Since it's difficult to nail down a narrow path, we will focus on Network status from the
Appliance point of view.Brand brand server
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It would also help to know something about the SIP protocol itself.
3.4.2 Getting started
In the voice exchange system's world, the network is usually divided in two legs:
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Check Operational Requirements for more details on pre-requisites.
3.4.3 Measure the Network
Given the above considerations, requirements and thresholds, we are going to measure the Infrastructure network. The following tests would perform the Infrastructure Network mostly, but we want to stress that a good Network test should involve the whole communication channels.
3.4.3.1 Measure the Packet Loss percentage
The first test to perform is about the Packet Loss percentage, as this can lead both to a Bad Quality communications (ie: gaps made of silence during the Secure Call) or worse: can't place calls at all. We are going to measure the packet loss percentage between the
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The test aims to simulate a network traffic with voice packets. As the voice packets are sent any 200 ms, the ICMP packets forged by the ping command "ping -i 0.02 pbx.mydomain.it" perform almost the same way. As a matter of fact we can read in the bottom line the test results. In particular we focus on the "2.4% packet loss", which is way over our basic threshold for a good Secure Call performance.
3.4.3.2 Measure the round trip average
The round trip measure has to be performed on the SIP protocol to have some debugging value. Thus we can desume the round trip by the client log, as a first step before deciding to choose a deeper investigation. Here we report an excerption from a client's log. When performing this analysis we look for a "REGISTER" request by our client and we note the time the message has been sent:
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So the roundtrip result is almost 1 minute, which is border line outcome by our basic threshold statements.
3.4.3.3 Consider the jitter
For this consideration, the output of "ping" command is useful, but it still needs a human eye pair to understand it correctly. The "ping" outcome about jittering won't be written down in an easy to read output line, instead it needs to be deduced by the time variation each line shows up.
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For a non jitter network communication we should have almost the same response time for each packet.
3.4.3.4 Following a SIP Session by User
If we are not sure about one client's behaviour and want to check out its network situation, then we first have to retrieve the user's name or peername (put some link here about this?). After that, we can check the client network history through the SIP Session Logs (link this to the SIP SessionLOG). Messages like "NETWORK_ERROR" or frequent "DISCONNECT" or "UNREGISTER", which might trigger a deeper analysis about the client status and its network performance.
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