Can I limit the speed of downloaded files (pictures, etc) to certain computers on my LAN

Hello,

I have a VOIP phone system.  It works well most of the time.  This morning while on the phone, the person on the other end started saying they couldn't hear me.  I later determined that my son was downloading a picture.  Can I put some kind of flow control on my router to give priority to the phone?  It is a Trendnet TEW633GR.

Thanks!
dennis2005Asked:
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Qlemo"Batchelor", Developer and EE Topic AdvisorCommented:
Higher level routers allow for honouring QoS or DiffServ markings. I.e. they priorisize VoIP traffic over HTTP or FTP traffic typically caused by downloads. However, if your Internet provider does not use that, VoIP may be suffering nevertheless.

Looking in a test report about that router, I see it seems to support QoS. However, the application or device you are using for VoIP has to support it, too, or (maybe) has to be plugged in a separate switch port. From the data I found about that router, I cannot tell.

I can only recommend that you have a look into the manual for QoS.
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DonrulezzCommented:
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dennis2005Author Commented:
Thanks to both of you for your suggestions.  I am reviewing the info at Speedguide.net.  There is a lot there.  So far I am not seeing how to limit speed.  I am totally new to routers.  Until recently, I only used switches.  
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Qlemo"Batchelor", Developer and EE Topic AdvisorCommented:
SpeedGuide is for optimizing your WAN connection. It doesn't give you means to control or priorize traffic.
There are some software solutions to restrict bandwidth, but they are "opt-in" based, i.e. each client has to run it, and each client can determine itself whether the limiter is used or not. And no limit on demand available, so you would restrict e.g. download all the time, whether VoIP traffic is flowing or not. Bandwidth limiting software is useful only in labs, e.g. for students, or other equally distributed limit demands.

Please try the QoS path. It shouldn't be difficult, if the VoIP device or software allows for using it. Maybe even the router recognizes SIP or similar VoIP traffic if you switch it on (but I don't believe in that).
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DonrulezzCommented:
VoIP,email,web browsing, downloading files, videos, music, etc are all competing for the same limited bandwidth space.
Combine all that with the fact that many ISP's don't always truly deliver the bandwidth they promise and now that big data pipe becomes a narrow pipe trying to squeeze at that data through at the same time.

1) Buy the fastest service you feel comfortable paying for.
But, I have heard stories where the ISP's don't always deliver all that extra bandwidth you are paying for.... so what can your do but give it a shot...

2) Of all the data services, VoIP is a "real-time" data service. It can't tolerate data bottle-necks like other data sources can. When your data pipe gets squeezed, VoIP suffers the most. VoIP can't tolerate dropped packets, or even delayed packets like other services can.

Sometimes we just can't have it all at the same time. Use discretion when you do heavy surfing and downloading. Try to make your VoIP calls during your data lulls rather than on the usage highs.

3) When you have a choice, use lower bandwidth VoIP Codec compression. Lower bandwidth, high compression Codecs are more bandwidth efficient and often provide better effeciency thant the high-qualigy Codecs.

For example, the standard high quality VoIP Codecs are:
G.711u/a (also known as PCMU/PCMA). I don't have exact figures, but typical bandwidth required for each leg of a G.711u VoIP call is about 64Kbps. But, when combined with call management bandwidth it can require up to 110Kbps... each direction. So, full duplex conversation could conceivably require about 220 Kbps. That may not sound like much for a high speed connection, but don't forget that 200+ Kbps VoIP data is competing with all the other data transfers occuring over your Internet connection.

When you have a choice, it is preferable to use a high compression Codec like G.729a. G.729a is rated somwhere around 32 Kbps bandwidth per leg (+ overhead)This Codec only uses about one-half the bandwidth of G.711u/a. So, just by using G.729a your could end up with better call quality just because your not dropping so many packets in the bottle-neck.
Most VoIP carriers will recommend you use G.729a in the name of reducing bandwidth. The difference in voice quality is minimal - and you may end up with less jitter and drop-outs.

If I recall, Vonage allows you to adjust your VoIP bandwith through your user admin web page. The feature is called "Bandwidth Saver". For the laymans sake, they just call the bandwidth settings as High, Medium, and Low (rather than confusing you with Codec standards names). In reality, your are just selecting different Codec compression algorithms. I think the low setting is something like the GSM Codec, which is basically mobile phone voice quality.
http://www.vonage.com/help.php?keyword=B&

4) All the above is the easy way to improve your VoIP call quality. Now for the technical side... and not so easy - QOS.

QOS - Quality of Service settings.
How to do this in detail is beyond what can be explained here.

QOS, in essence, is the prioritizazation of VoIP packets.
When many kinds of data are competing for the same space within a limited data pipe, QOS acts like the traffic cop directing the flow of traffic. With QOS, you would want to establish that VoIP devices get priority of data traffic over other data flows, like email, web, and file transfers to your PC.

The trick with QOS is that you need a router that has QOS priority features. Then you have to configure the QOS setting in the router to give the VoIP adapters "Highest Priority" over other data traffic. How this is implemented may vary from Router to Router.

Take for example, I have a Linksys WRT54G NAT/Router and a PAP2T-NA VoIP adapter (ATA). The ATA is assigned an IP Address by the router. The ATA also has a unique MAC Address. The key here is that I had to configure the WRT54G to priortize data flow to the specific IP address (and ports) as required by my ATA. Now, once QOS is correctly configured and routed, my ATA VoIP call data gets top priority over competing data on the data pipe flowing through my Router. (Some routers may be able to prioritize based on MAC address of the ATA too)

Some ATA's are combo boxes that combine the NAT/Router and a VoIP adapter in one box. The Linksys SPA2102 and SPA3102 are examples. I haven't used those boxes, so I don't know how they are set-up for QOS, by default. I would hope they give all data to the built-in ATA top priority data flow.

So, in a nut shell, that's the best advice I can give here.
Hope it helps.
(It may not solve your problems, but may give you a better understanding of whats happening, or needs to be done)
Source(s):
http://www.cqvoip.info
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Qlemo"Batchelor", Developer and EE Topic AdvisorCommented:
This was pretty much elaborated. I've to add that VoIP compression might add a small lag depending on the efficiency of the compression algorithm implemented, and results in loss of speed quality.
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dennis2005Author Commented:
Thanks to both of you for your input.  I have been pouring over the manual for the Trendnet Router.  It does have QoS capabilities.  I am trying to determine what I need to do to implement them properly.

I do GREATLY APPRECIATE HELP FROM BOTH OF YOU!
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