AFFECTED FOR VOICE QUALITY ? - READ THIS PAGE AND CHECK YOUR BANDWIDTH
CHECK YOUR BANDWIDTH SPEED
HERE->>>
Download Speed: A rating of how well your Internet Service Provider transfers data across the Internet. The higher the percentage the better your connection is. Please note, this value does not represent the quality of your APTECH service.
Upload Speed: Upload is the measure of how fast content is delivered from your computer or local area network to others on the Internet. Please note, this value does represent the quality of your APTECH service.
You can use this table to estimate the bandwidth required to support a given number of lines (voice paths) across your broadband connection or for reverse calculations you can view how many lines your current bandwidth can support. Two compression algorithms are assumed called Codecs. For example, if you require 5 lines @ G.729a compression, you would need a minimum of 120kbps to support it. In reverse, if you have 320kbps bandwidth, you could support 4 lines at G.711 or 13 lines at G.729. This table is to be used for estimation only, as there could be many other factors unique to you, that are not taken into consideration.
Using the Internet on your Computer while you are
talking on your APTECH Phone?
You can talk and use your internet at the computer at the same time, there is no
conflict between them, you just need to check your available bandwidth, for
example, let's say you have a DSL service with 6.0MB download and 512kb upload
that is the Internet bandwidth normally provided to businesses by the phone
company, if you have 4 lines and you are talking on the 4 at the same time, you
are taking about 96kb of your upload speed using the Codec g729 to compress the
voice, if you use G711 you are taking 320kb, out of the 512kb there is no
problem in theory, problem comes when you are using the four lines and somebody
else in your office wants to watch TV over the Internet and/or send or receive
heavy files on the email. Normally music does not take as much bandwidth as
video streaming, intensive video in your connection may kill the bandwidth
available, in this case you are exceeding the capacity of your bandwidth and for
this reason you can hear choppy voice. Solution for this problem is to
setup your router for QOS (Quality of Service), in the case we are talking about
you must reserve a minimum of 120kb for voice priority.
|
Number of |
Codec |
Codec |
| 1 | 24 | 80 |
| 2 | 48 | 160 |
| 3 | 72 | 240 |
| 4 | 96 | 320 |
| 5 | 120 | 400 |
| 6 | 144 | 480 |
| 7 | 168 | 560 |
| 8 | 192 | 640 |
| 9 | 216 | 720 |
| 10 | 240 | 800 |
| 11 | 264 | 880 |
| 12 | 288 | 960 |
| 13 | 312 | 1040 |
| 14 | 336 | 1120 |
| 15 | 360 | 1200 |
| 16 | 384 | 1280 |
| 17 | 408 | 1360 |
| 18 | 432 | 1440 |
| 19 | 456 | 1520 |
| 20 | 480 | 1600 |
| 21 | 504 | 1680 |
| 22 | 528 | 1760 |
| 23 | 552 | 1840 |
Bandwidth:Lines Assumptions:
Coding algorithm: (G.729A (CS-CELP) 8kbps compression + overhead)
Coding algorithm:(G.711 (PCM) 64kbps uncompressed + overhead)
Packet Duration: 20 milliseconds
What could be the problem of the Voice Quality using our APTECH Service?
There can be lots of reasons for poor VoIP call quality. Here's a few of them.
Poor call quality is not usually the fault of the VoIP service provider (VSP), although it can be. Lets not forget that for VoIP to VoIP calls that if you connect via a VSP that the rest of the call is not routed through their network. For calls to PSTN numbers (regular landline and mobile numbers) the quality of the PSTN leg, wherever that might be routed through and end up, is not within the control of the VSP, unless they have chosen to route/terminate their calls via an unreliable telco.
There are other issues which can affect the quality of a VoIP call:
1. The quality/latency of your connection to the web, which is down to your choice of ISP. (VoIP over a non-line of sight, non-WiFi hotspot, broadband wireless connection can have the same issues as VoIP over any other type of connection, as well as particular issues that are mentioned in point 9.)
2. The quality/latency of the other person's connection to the web if you are doing VoIP to VoIP, which is down to their choice of ISP.
3. Your choice of codec, as some codecs will give higher fidelity sound than others, although none of the commonly used codecs will give lousy sound quality, all other matters being equal. (The codec that you select on your softphone or VoIP hardware device must match one that is supported by your VSP.
4. If you are using a softphone or an ATA, whether or not running too much other stuff on your computer thereby slowing down its processor too much. Similarly, whether your computer's processor is fast enough.
5. If you are using a softphone or an ATA, whether or not you are using one of the better sound quality/echo options of using a noise canceling microphone, a USB headset or USB phone, rather than a regular headset that plugs into your sound card.
6. Whether or not you have enough bandwidth available to make a good quality call. That will depend on your choice of codec and whether or not you are running other applications/computers which are accessing the web via the same internet connection which you are using for the call. QOS software/firmware can prioritize your VoIP traffic to ensure you have enough bandwidth available for your calls; also some ISPs support QOS over their networks. Similarly you can simply ensure that no other application/computer is accessing the web while you are making your calls, thereby achieving the same result as QOS.
7. Whether or not you have echo cancellation firmware/software, and whether or not you use it. Similarly, whether or not the speaker volume on your IP Phone / regular phone plugged into an ATA / USB phone / ATA / headset or softphone is too high, or if it the volume is too high in the ATA, IP Phone or softphone settings, thereby resulting in feedback induced echo to the person to whom you are talking.
8. It is wise to not connect the computer running the softphone or ATA to a hub. If possible connect them directly to a LAN switch port on your router. Hubs provide a 10mbit shared medium, and are generally unsuitable for VoIP. However, if a switch has not been specifically configured to identify and prioritise VoIP packets using QoS, latency and jitter may be encountered, and in some extreme circumstances may introduce up to a second in delay when saturated with traffic.
9. Good quality VoIP over a non-line of sight, non-WiFi, wireless broadband connection, can often be achieved. (Many people have reported having good quality VoIP using iBurst; Unwired appears to be marginal due to relatively high latency; CDMA/GPRS connections appear to have too much latency, and not enough information is available to comment about the viability of good quality VoIP on WCDMA/UMTS connections.). The most important issue is to ensure the signal strength and quality are maximised to ensure that packet loss, or frame error rate, is zero or very little. Maximising the signal strength and quality can also reduce latency in the connection and reduce lag during VoIP conversations. There is also one tweak that can sometimes be beneficial: some softphones and VoIP hardware have interfaces where the user can alter the size of the VoIP packets. Sometimes altering their size can overcome, to some extent at least, the small sound drop-outs due to packet loss triggered by the higher latency and variations in latency which are common in non-LOS wireless broadband connections. The recommendation is to change the packet size one step at a time to see if there is improvement or not.
