Reddit reviews GE DSL Phone Line Filter (76249)
We found 2 Reddit comments about GE DSL Phone Line Filter (76249). Here are the top ones, ranked by their Reddit score.
Shields Phone Lines From Digital Noise & Interference Caused By DslOne Filter Needed For Each Telephone Device (Telephone, Fax Machine Or Answering Machine) Sharing A Dsl LineWhiteShields Phone Lines From Digital Noise & Interference Caused By DslOne Filter Needed For Each Telephone Device (Telephone, Fax Machine Or Answering Machine) Sharing A Dsl Line
TL;DR- yeah there are ways to do this but I don't necessarily recommend them. If the heart monitor is life-safety equipment, and a failed dial-in could put your wife's health at risk-- please don't attempt any of this, just stay with the cable company line. Saving $20/mo isn't worth losing your wife over.
Okay a few things to understand first.
The RJ11 port on your laptop modem is an FXO port (just like the monitor), and is physically incapable of providing a phone line for your wife's heart monitor.
There are MagicJack type gadgets that DO provide an FXS port, or dedicated devices called ATAs (Analog Telephony Adapter) that have Ethernet on one end and FXS on the other. You'll need something like that to make this all work.
If your VoIP service uses a lossy codec, then it will not work at all with your heart monitor.
Some VoIP services let the user select which codec to use- the only codec that will work with modems is G.711 uLaw/aLaw.
Most VoIP systems transmit a voice packet approximately every 20ms. Jitter can cause a delay between packets (causing the receiving end to run out of audio to play), or cause packets to be delivered in a bunch (in some cases causing one to be skipped). This causes little problem for most VoIP as 20ms of lost audio won't interrupt a conversation. However modems cannot deal with jitter- modems are looking for specific sound waves to happen or not happen at specific points in time. If that sound happens later or earlier than expected, it can disrupt the data transmission.
Some VoIP systems have a 'jitter buffer'- to guard the audio against jitter, each end buffers about 100ms worth of audio. That way packets can come in whenever they get in, but the audio is played correctly out of the buffer.
Modems can deal with latency, but not jitter. So you need to make sure your VoIP system has a relatively fixed jitter buffer for reliable data transmission.
As such, I'd suggest a hard think about what the benefit of this system is (lower cost presumably), but more importantly, what's the consequence if it fails. If a failed connection from the heart monitor could mean health consequences for your wife, then I'd strongly suggest scrapping this idea and sticking with the cable company phone service. This may be a fun project and might save you a few bucks, but that's not worth losing your wife over.
That all said- If you want to do this, I don't think the laptop is the way to go. I suggest purchasing a basic ATA, such as a Cisco/Linksys PAP2T-NA or Cisco SPA112. Both are available from Amazon.
Then you'll want a basic VoIP service. For what you're doing (outbound calls only), I suggest http://voip.ms . You don't even need to assign your ATA a phone number, so there's no monthly fees, it will just charge a cent or two every minute each time your heart monitor dials out. This will reduce your monthly spend to probably well under $1 (that's not a typo). Note that in this configuration, there will be no 911 service on this system.
However, configuring this is non-trivial. The Cisco ATAs have approximately 912743832487 options to configure, many of which will affect operation with your heart monitor. Your general process will go like this- Setup voip.ms, get the SIP credentials, and find a guide to configure the ATA for voip.ms (their support page should have one). Then for the line/port that you're using, turn on jitter buffer, type is fixed (not adaptive), length is medium. Enable fax mode for always (not auto detect). Disable call waiting. Disable echo cancellation. Enable make call without registration. Set codec to be ONLY G.711 uLaw.
Now go in your router setup. Look for Quality of Service (QoS) or traffic priority or something like that. Not all routers have this feature. Prioritize the traffic from your ATA device to the highest level. This prevents a big download from interrupting the heart monitor.
Now get yourself a DSL filter. Plug it into the ATA. The filter removes non-audio frequencies and can make data over VoIP more reliable.
Next plug a normal analog phone into the DSL filter and make some test calls. Expect there to be a slight delay (due to jitter buffer) and echo (due to no echo cancellation). That's fine, they don't affect faxing. Aside from delay and echo, voice quality should be very good with no dropped syllables or anything like that.
Finally plug the heart monitor into the DSL filter and force it to phone home. Do this 3 or 4 times. It should be able to successfully dial in every time.
Hope that helps! Feel free to ask if you have any questions...
http://en.wikipedia.org/wiki/DSL_filter
http://www.amazon.com/GE-Phone-Line-Filter-76249/dp/B002HEQD8K/ref=sr_1_1?ie=UTF8&qid=1409947828&sr=8-1&keywords=dsl+filters
I remember when I got my DSL sercie back in 99 the phone company sent me like 10 of these guys, if I ever changed phones and forgot to put the filter back in place it would make my dsl connection drop or act weird.