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Trixbox - The Easy Tutorial - Network Link Quality

Trixbox Link Quality
Last Change : Jun 30 2010 french flagenglish flag


Tool
Install
Ergonomy
Forum



Details What is Trixbox ?
Screenshots
Prerequisites
Configurations
Softphones
---- Statistics ----
Voice protocols
Link quality




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The advantage of VoIP is that it uses existing data network links and thus saves money. However, the network link's quality must be sufficient otherwise the audio would be low quality or broken.
This has been a major problem since the late '90s when VoIP began to take off, the network link's quality was often too bad leading to unhappy phone users.

VoIP remains a cost-saving solution and represents the future compared to voice over the public switched telephone network (PSTN) but has to be prepared carefully before being implemented, the network links' quality being very important.



A network link quality depends mainly on the three following issues:

1. Latency or delay
2. Jitter (latency variation)
3. Datagram loss



1. Latency.

Latency, also called delay, is the one-way time taken for a packet to travel across the network between two hosts.
It should not be confounded with the RTT (Round Trip Time) or response time which measure latency in both directions.

VoIP recommended value (phone - PBX): Less than 150-200ms
If the recommended value is exceeded, the people having a phone conversation will have to wait a long time before hearing what each other say. This will produce problems because after you have spoken, you do not know if the other person hasn't spoken or if you have to wait for an answer.

2. Jitter

The jitter is basically the latency variation and does not depend on the latency. You can have high latencies and a very low jitter.

VoIP recommended value (phone - PBX): Less than 5ms
The jitter will affect order of the packets' arrival. To solve this problem, the VoIP partners have a buffer to reorder the packets in case of need. The buffer size cannot be too high as otherwise this would slow down the audio communications.
In the event of high jitter, the buffer will be full and thus some packets will be discarded which concretely means that some conversation words or parts of words won't be received.

3. Datagram loss

The datagram loss is frequently displayed as a percentage. It shows the quantity of packets lost during their travel between two hosts.

VoIP recommended value (phone - PBX): Less than 1%
Audio is transported inside the RTP protocol. As RTP is located over UDP in the OSI layers model, it does not offer any guaranteed delivery mechanism such as TCP's. In other words, when a packet is lost during its travel, it will not be retransmitted.

This clearly shows the importance to keep a low datagram loss percentage. The loss of only one packet will disrupt, even if only very briefly, a phone conversation.
If this percentage is too high, the conversation will become unintelligible or even broken.



Tools:

Five free and open source tools provide a way to measure the network link's quality.

Ping:
Ping is used to test IP connectivity. It provides RTT and datagram loss information.
It is extremely simple to use and is installed by default on nearly all operating systems.
Unlike the two next tools IPerf and D-ITG, you don't need to have the destination machine on hand to install and configure the tool.

IPerf:
IPerf is primarily used to measure the available bandwidth between two hosts running IPerf. It is also possible to measure the jitter and the datagram loss.
Check the IPerf tutorial for a detailed help.

D-ITG (Distributed Internet Traffic Generator)
As with IPerf you have to set D-ITG on the both the source and destination machines.
It has a graphical interface and is extremely powerful. It will fulfill all your network measurement needs!

Wireshark (former Ethereal): (See full tutorial)
The best network analyzer. It provides a lot of informations about VoIP such as jitter.

WANem: (See full tutorial)
A tool which is located between two hosts such as a phone and a PBX to simulate a specific quality of a network link. Settings such as latency, bandwidth, datagram loss and jitter are available.
As WANem is "in the middle", routing must be set on the two test machines to force the traffic between each other to pass through WANem.

Summary:


Measurement:

Simulation:
Latency:
D-ITG

WANem
RTT:
Ping / D-ITG

WANem
Bandwidth:
IPerf / D-ITG

WANem
Jitter:
IPerf / D-ITG
Wireshark
WANem
Datagram Loss:
Ping / IPerf
D-ITG
WANem





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