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Mike AFlag for Canada

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VOIP Phone - Residential Internet router settings

is there a standard way to set up a Residential Internet service for VOIP Phones?
the quality of the voice is poor and i was told there was a way to tweek or configure the home router to prioritize voice traffic
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hypercube
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The only thing that I've run into in doing things like this is that in the router, SIP ALG needs to NOT BE ENABLED for most routers and most VOIP providers.  They may be able to tell you - then again, maybe not.

That said, I don't know what might happen re: anomalies if it's enabled.  Poor voice quality maybe.  Not working at all maybe.  Provider-dependent maybe.
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Dr. Klahn

If the VOIP box supports QoS (Quality of Service), some routers can be set to allow preference to traffic to the VOIP box.  But this does no good unless there is a potential for resource competition, i.e. a large amount of traffic also trying to use the router at the same time.  If there is no other traffic, or only a small amount of traffic, QoS won't make any difference.

But if this is your first experience with VOIP, do realize that VOIP uses highly compressive codecs, voice bandwidth filtering and relatively low sample rates (only 8 KHz.)  The result is that the quality tends to be lower than that achieved over traditional copper lines and POTS (Plain Old Telephone Service) analog phones.

There is a lot that can go wrong or being suboptimal together creating a trainwreck.

Very asymetric (20/2 DSL)  links can hurt.  (combined with large buffers).   a thing known as buffer bloat, something a LOT of domestic routers sufferfrom.

The uplink of such connections are filled with ACK packets for the downloads.

Heavy downloading will hurt VOIP traffic.   

Oversubscription will hurt. (too many subscribers on the feed links to the area). 

With DSL if you can have CELL loss (not packet loss, but only a part of it)  it will hurt as every cell loss, also means a packet loss), a cell is about 30 bytes of data,

so you need at least 2 cells for an empty UDP packet.  Cells are not retransmitted.   If a DSL link has a bad quality then VOIP will be hard.

(Voip packet are about 20ms of sound each). 

Most SIP ALG's  are doing the wrong thing(tm)  in various creative ways. 

Inconsistent NAT hurts...


Thing that can help: 

If you can do bandwidth management prioritize UDP RTP packets  or at least limit all other traffic. 

A tool like wondershaper (Linux Router Project from 2000 -era)) has a description on how it can help. (only if you run your OWN router).

Cell loss can be alleviated by the ISP if their end (DSLAM) can do autotraining which will limit the speed if there is to much cell loss creating a more stable link.

Every half decent VOIP PBX can handle FENT (Far End Nat Translation) to cope with NAT.

Creating port forward for a phone (doable for low numbers) with fixed ranges per phone will help.


If you have many phones look for on premises PBX for all internal traffic. (otherwise you need to entertain 2 connections to the PBX of the Voip Provider).

(3CX, FusionPBX) can be used for free when there are not a lot of phones.

is there a standard way to set up Residential Internet service for VOIP Phones?
the quality of the voice is poor and I was told there was a way to tweak or configure the home router to prioritize voice traffic

(I'm guessing you're from the US and your bandwidth are above 50/50mbps) - Now, the only standard for me is selecting which carrier you are using for SIP trunk. If your carrier is not reliable with its connection you may face choppiness. jitter, among others.

Now, a few questions for you, are you always downloading? does anyone at that place is BitTorrent something? all that count.

VOIP does not consume that much bandwidth, we are talking about 64kbs at best. It all depends on what codec is you and your provider are using.
You need to find out what's the bandwidth on your residential environment.

You need to determine is there's no latency between your customer and the SIP server. Latency cannot go beyond 120 ms, in my case. That means how long a packet takes from one end to another.

The connection from the server to the client should be always like this (less 50ms Latency / 20m jitter - codecs G711ulaw (if you're from the US) 64kbs).

Don't bother trying to fix something that you can't fix. It _COULD_ be a priority issue, but I think it's probably not.

Shutdown or disconnect ALL internet enabled devices. Only the VOIP phone stays online obviously. Make a call. Call quality still bad? Don't bother figuring out priority (some modems don't even have this option), but get on the phone with the support dept. They need to help you figure it out (IT manager in a big firm could do it, but since you're a consumer, the VOIP helpdesk should guide you through, do real monitoring and give you hints and tip if it's an issue on your side (voip phone config), or fix if the problem on their side).


On the other hand, if the call quality is suddenly crystal clear, sure, go through all available tips posted above.

Who is the provider for your internet and make model if the device where their connection is?
Some provider provided rpurers, including the ones with wireless might have SiP ALG enabled, and unmenagable.
Often, the vendor through whom you get services has a list, guidance depending on the provider, routers they support.
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