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<blockquote data-quote="Pheasant Surprise" data-source="post: 6573205" data-attributes="member: 1103"><p>...assuming you’re not being facetious J B <img src="data:image/gif;base64,R0lGODlhAQABAIAAAAAAAP///yH5BAEAAAAALAAAAAABAAEAAAIBRAA7" class="smilie smilie--sprite smilie--sprite7" alt=":p" title="Stick Out Tongue :p" loading="lazy" data-shortname=":p" />...</p><p></p><p>Voice traverses the internet like any other data, as the internet is a giant “packet switched” data network. The senders “codec” chops the voice signal up into little packets of data and sends them on their merry way to navigate. These data packets are then reassembled at the remote end by the receivers codec to form (hopefully) intelligible speech.</p><p></p><p>However as the path between the sender and receiver can be quite inconsistent and variable due to the dynamic nature of the internet, including bursts of traffic, and the variety of ways folks connect into the core internet network, it can introduce various delays. This directly affects the quality of the service (so called QoS) for voice.</p><p></p><p>Delay is a measure of how long it takes to send a packet (series of them usually) and to get the associated response. It’s measured in milliseconds. When you “ping” a server you are measuring this aggregate delay. For decent quality voice over IP connection you’d want measured ping times to ideally be less than about 50 ms. Lower is better and higher is worse. High ping times can result in a echo effects.</p><p></p><p>Jitter is the measure of the variation in delay between the arrival of different packets from sender to receiver. Think of the data packets as cars driving in a multi-lane road. If the cars were even across the lanes then there would be no jitter. On the other hand if there is a big spread of distance between the cars in lanes A, B and C then they would cross an imaginary line at different times. This is jitter, it’s a measure of this spread in arrival times of the various packet.</p><p></p><p>Again for voice we want the jitter (variation in delay) to be as low as possible. High jitter times result in a range of distortion to the voice service. Ideally a jitter time less than around 15 milliseconds.</p><p></p><p>The final thing with voice quality is packet loss. Hopefully that’s self explanatory, we’re aiming for less than 1% packet loss for voice. Packet loss is really bad as it results in drop outs and gaps in the conversation.</p><p></p><p>if you want to read more, here are some good links.. </p><p></p><p>[URL unfurl="true"]https://www.voip-info.org/qos/[/URL]</p><p></p><p>[URL unfurl="true"]http://www.ciscopress.com/articles/article.asp?p=357102[/URL]</p></blockquote><p></p>
[QUOTE="Pheasant Surprise, post: 6573205, member: 1103"] ...assuming you’re not being facetious J B :p... Voice traverses the internet like any other data, as the internet is a giant “packet switched” data network. The senders “codec” chops the voice signal up into little packets of data and sends them on their merry way to navigate. These data packets are then reassembled at the remote end by the receivers codec to form (hopefully) intelligible speech. However as the path between the sender and receiver can be quite inconsistent and variable due to the dynamic nature of the internet, including bursts of traffic, and the variety of ways folks connect into the core internet network, it can introduce various delays. This directly affects the quality of the service (so called QoS) for voice. Delay is a measure of how long it takes to send a packet (series of them usually) and to get the associated response. It’s measured in milliseconds. When you “ping” a server you are measuring this aggregate delay. For decent quality voice over IP connection you’d want measured ping times to ideally be less than about 50 ms. Lower is better and higher is worse. High ping times can result in a echo effects. Jitter is the measure of the variation in delay between the arrival of different packets from sender to receiver. Think of the data packets as cars driving in a multi-lane road. If the cars were even across the lanes then there would be no jitter. On the other hand if there is a big spread of distance between the cars in lanes A, B and C then they would cross an imaginary line at different times. This is jitter, it’s a measure of this spread in arrival times of the various packet. Again for voice we want the jitter (variation in delay) to be as low as possible. High jitter times result in a range of distortion to the voice service. Ideally a jitter time less than around 15 milliseconds. The final thing with voice quality is packet loss. Hopefully that’s self explanatory, we’re aiming for less than 1% packet loss for voice. Packet loss is really bad as it results in drop outs and gaps in the conversation. if you want to read more, here are some good links.. [URL unfurl="true"]https://www.voip-info.org/qos/[/URL] [URL unfurl="true"]http://www.ciscopress.com/articles/article.asp?p=357102[/URL] [/QUOTE]
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