Understanding Your Network Quality of Service (QoS)
The quality of service (QoS) refers to several related aspects of telephony and computer networks that allow the transport of traffic with special requirements. In particular, much technology has been developed to allow computer networks to become as useful as telephone network for audio conversations, as well as supporting new applications with even more strict service demands. In packet-switched networks, quality of service is affected by various factors, which can be divided into “human” and “technical” factors. Human factors include: stability of service, availability of service, delays, user information. Technical factors include: reliability, scalability, effectiveness, maintainability, grade of service, etc.
Many things can happen to packets as they travel from origin to destination, resulting in the following problems as seen from the point of view of the sender and receiver:
Due to varying load from other users sharing the same network resources, the bit rate (the maximum throughput) that can be provided to a certain data stream may be too low for realtime multimedia services if all data streams get the same scheduling priority.
The routers might fail to deliver (drop) some packets if their data is corrupted or they arrive when their buffers are already full. The receiving application may ask for this information to be retransmitted, possibly causing severe delays in the overall transmission.
Sometimes packets are corrupted due to bit errors caused by noise and interference, especially in wireless communications and long copper wires. The receiver has to detect this and, just as if the packet was dropped, may ask for this information to be retransmitted.
It might take a long time for each packet to reach its destination, because it gets held up in long queues, or takes a less direct route to avoid congestion. This is different from throughput, as the delay can build up over time, even if the throughput is almost normal. In some cases, excessive latency can render an application such as VoIP or online gaming unusable.
Packets from the source will reach the destination with different delays. A packet's delay varies with its position in the queues of the routers along the path between source and destination and this position can vary unpredictably. This variation in delay is known as jitter and can seriously affect the quality of streaming audio and/or video.
When a collection of related packets is routed through a network, different packets may take different routes, each resulting in a different delay. The result is that the packets arrive in a different order than they were sent. This problem requires special additional protocols responsible for rearranging out-of-order packets to an isochronous state once they reach their destination. This is especially important for video and VoIP streams where quality is dramatically affected by both latency and lack of sequence.
Establishing the cause of poor Quality of Service (QoS)?
Simple Qos Testing
Advance QoS Testing
- Open the Start Menu
- Type "cmd" into the search bar, press Enter (this will open a command window)
- Type "ping 220.127.116.11 -t" into the command windo
- 18.104.22.168 is the Real Time Protocol (RTP) server that streams audio to your phones. You should see no packet loss and your ping time should average 20-30 milliseconds.
- Press "Ctrl + C" after about 60 seconds of monitoring to see your results.
Submitt results to http://support.4c-comm.com, if speed and quality are poor contact your Internet Service Provider (ISP) and have they begin troubleshooting the challenge. If 4C is your ISP, 4C will initiate the troubleshooting process.
If Quality of Service is not Network Related
- Generate a Call Log
- Time & Date,
- Origination Number
- Termination Number
Submitt Call Log to http://support.4c-comm.com (You should send 1-3 logs to insure a pattern discovery on your system)