What are the factors impacting Quality of Service?
From Bandipedia
Defining standards to measure voice quality in conjunction with availability detection can enhance VoIP. As a result, a well-managed network plays an essential role in keeping a VoIP system up and running properly. The following sections describe two important classifications that should be considered when defining quality standards for VoIP, latency and jitter.
Latency
From a PSTN viewpoint, latency can be described as the amount of time it takes a caller’s voice to reach the listener’s ear. The larger the latency value, the more likely it is for a lack of synchronization between speakers. Although a larger latency value does not necessarily degrade the overall sound quality of a call, it may promote hesitation in the participants’ interactions during a call. Industry standards suggest that end-to-end latency should be kept to a 150 ms maximum. Network operators must take into consideration the various causes of latency delay to ensure the delay remains below 150 ms.
- “Packetization†delay: amount of time it takes for endpoints to create packets used in voice services. The larger the packet size, the greater the amount of time it takes to fill. While packetization delay is governed by the CODEC standard being used, nominal operation should not exceed 30 ms.
- Serialization delay: amount of time it takes to serialize digital data onto physical links of interconnecting equipment. Generally speaking, the faster the media, the lower the latency.
- Propagation delay: amount of time it takes a signal to traverse the length of a conductor. While there is always going to be some level of propagation delay, it is only an issue when the signal travels a great distance.
- Queuing delay: amount of time a packet remains buffered in a network element while awaiting transmission.
- Packet forwarding delay: amount of time it takes for a network device to buffer a packet and make the forwarding decision. This delay is variables and depends on the function and architecture of the networking device.
Jitter
Jitter is the measure of time between when a packet is expected to arrive to when it actually arrives. Queuing variations caused by dynamic changes in network traffic loads is a major cause of jitter. Packets that take different equalcost links not physically the same length as other links is another cause and can disrupt VoIP quality. Play-out buffers shield media gateway packet streams so that voice quality is not as affected by packet jitter. While jitter cannot be completely eliminated, play-out buffers can help minimize its effects; thus increasing the overall voice quality of a VoIP transmission.
What do you think about this page?
Comments are appreciated and assist in building this site.
We welcome comments, question, and suggestions in the following manner:
1. Click here to leave a comment.
2. Send an e-mail to wiki@bandwidth.com
3. Call 919-297-1069 and record your comments
