QoS is one of the important factors for any successful VoIP application. QoS mechanism is getting advanced each year, and the mechanism works better from smaller LANs to bigger organizations. In networking concept, the term quality refers to various factors. For VoIP, the term quality refers to the voice that is clear to speak and the voice that is clear to listen, without any additional noises. Additionally, quality also depends on packet loss, jitter (both positive and negative jitter), and echo. In networking concept, the term service also refers to various factors. For VoIP, the term service refers to the communication facilities that are offered to the customers.
Different levels of data priority
The QoS setting on the router can be different in the importance levels for the data such as important, best-effort, high, medium, low, and critical. These levels of importance can be configured depending on the router types. The popular type of QoS tagging in the router is the Ethernet port, endpoint’s MAC address, and UPD or TCP port.
Enabling QoS in VoIP
The QoS setting enables the router to prevent unwanted noises and increases the quality of the VoIP connections. By providing the priority to the voice and voice traffic, the router helps to prevent other services having the similar priority level, ensuring a proper data flow maintenance as well as a proper voice connection. In general, the QoS is a service that prioritizes certain important packets to reach the destination and delays the packets that are least important. These prioritized packets reach the destination as soon as possible. If the router is not enabled with QoS, then there is no concept of prioritizing any important packets, and all the packets pass the router through a concept called First In, First Out (FIFO). When a router is able to calculate the total data it can receive, then the router can control the data traffic through queues, by sending the high priority packets first and delaying the low priority packets.
To understand better on what QoS setting can achieve in VoIP, it is necessary to know the location of the router where the QoS is implemented. The location of the router will be in the LAN as the gateway device and will be located after the cable modem or the DSL, but not after the VoIP ATA. A proper QoS setting on the router can eliminate unwanted noises that are caused by other applications taking the required bandwidth. The other applications typically refer to email, games, and any other application that consumes high bandwidth.
QoS setting that cannot do in VoIP
It is also necessary to know what will not work on the QoS setting. The QoS setting of the router that is configured on a network will not have any impact on the performance from the Internet Service Provider (ISP). In general, the ISP has already set the download and upload limits, and it depends on the ISP’s level of service. The ISP’s level of service determines the bandwidth of both download and upload limit, and some of the speed test measurements can help to determine the limits. If the determined download limit and upload limit is not received for at least 80% of the mentioned limits from the service provider, then it is recommended to check the service provider’s troubleshooting methods or documentation, before trying for any other implementation.
QoS for VoIP
The following categories explain the importance of QoS for VoIP:
Overview of QoS for VoIP – For a successful VoIP in the Public Switch Telephone Network (PSTN) telephony services, it is necessary to receive the continuous voice transmission without any other disturbances. Similar to any other real-time applications, VoIP is also delay-sensitive. In order to receive a good VoIP transmission to the receiver, the packets required for voice must not be dropped, must not be delayed or suffer from jitter. A jitter can be classified into positive jitter and negative jitter. If the ideal time of a packet to reach the destination is more than the normal time, then the jitter is called the positive jitter. If the ideal time of a packet to reach the destination is lesser than the normal time, then the jitter is called the negative jitter. In general, the jitter should be neither positive nor negative, and it must be ideal for the VoIP transmission. The QoS for VoIP provides better network service only from the following features:
o Dedicated bandwidth support
o Improvement in loss characteristics
o Management in network congestion
o Shaping while network traffic
o Traffic priorities across the network
Sufficient bandwidth – Before applying any QoS setting, it is recommended to make sure the network bandwidth to support any real-time voice application or traffic. In general, it is assumed that out of 100% packet, 20% packet is dropped. This assumption is considered only if there is no other traffic flow is available on the network. After ensuring the proper bandwidth for the voice application, further processes could be taken to guarantee that the voice packets have certain priority to reach the destination.
Queue mechanism for QoS – After the data is configured based on the QoS requirements, it is required to provide priority service and bandwidth guarantee through a process called queue mechanism. One of the popular queuing mechanisms is the Low Latency Queue because this mechanism is flexible and easier to configure. The important factor of the Low Latency Queue is to configure certain minimum bandwidth priority to VoIP applications and minimum guaranteed bandwidth priority to all other required applications. If everything is configured as per the requirement, then the priority queue where the traffic goes never exceeds the desired or configured rate.
Fragmentation and interleaving – VoIP transmissions are always delay-sensitive, and the VoIP packets are necessary to be inserted between the fragments. If everything is configured correctly, with proper voice traffic prioritizing, at times, the priority queue gets empty, and the packets from the non-priority service are taken. If the packets from the priority queue arrive at the output queue while these packets get services, then certain packets from the priority queue will have to wait for a certain time before reaching the destination. For example, in a 64-kbps link speed and the maximum transmission unit size of 1500 bytes, the calculation of the delay is approximately 187.5 milliseconds. In this situation, this means that a packet from the priority queue has to wait for approximately 187.5 milliseconds to reach the destination. Generally, a packet from the priority queue reaches the destination within 20 milliseconds. In certain cases, this delay is considered as a jitter and a delay of about 187.5 milliseconds could be more and not acceptable. It is necessary to have a mechanism to ensure the transmission unit should reach the destination in less than 10 milliseconds.
Traffic shaping – The term traffic shaping refers to a QoS mechanism to send traffic in a shorter time at a configured rate. This mechanism is generally used in Frame Relay environments, in which the interface clock rate is not same as the configured bandwidth.
Things to do before setting QoS for VoIP
Whenever there is a disturbed or unwanted noise in the VoIP application, try these steps before setting QoS for VoIP. These steps would help to reduce the issues.
Try to talk on a VoIP phone without any IP phones or computer and check if the issue is still persisting. If the issue continues to persist, then try to do a G729 codec setting. Some providers refer G729 codec as a bandwidth saving setting.
Make sure that any node or nodes connected to the network is clean from hazardous spyware. It is recommended to use a reputed anti-virus application to detect the spyware.
If any external network monitoring tool is configured for the network, then it is recommended to check for any issues with the VoIP service and generate a report for at least one-week period.
Based on the information mentioned here, it is a critical requirement to have the QoS settings configured for the VoIP applications. Apart from QoS settings, it is also necessary to have a network-monitoring tool configured on the network to monitor the performance as well as the stability of the VoIP applications.