(Part 2) Top products from r/VOIP

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.

1. Phone ports. There are two types of phone ports- FXO and FXS. FXO ports connect to a line, FXS ports provide a line (and battery voltage / ringing / etc). So the port on your heart monitor is an FXO port, the port on your cable modem is an FXS port. Note that FXO/S refers to the electrical configuration of the port, not the physical construction of the port.
   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.
   
   
2. Codecs. When VoIP services transmit audio over the Internet, that audio is encoded using a codec. There are a whole handful of codecs used by VoIP, all with various benefits and limitations. However the vast majority of VoIP codecs are 'lossy' codecs- that means the audio that comes out is not exactly the same as the audio that goes in. In order to save bandwidth, a 'lossy' codec throws away some of the audio data. They're designed to work in a way that humans won't notice much or at all, but a lossy codec is incompatible with modems (including faxes and heart monitors) because modems depend on sound to be transmitted PRECISELY.
   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.
   
   
3. Jitter. Latency is the Internet delay between when you transmit a packet in one place and when that packet is received in another place. The PING tool in Windows measures latency. Jitter is changes in latency- if your latency between you and your VoIP service varies from 20ms to 25ms, you have 5ms of jitter.
   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.
   
   
4. Reliability. As you can hopefully see, getting data transmission to work over VoIP is a bit harder than getting voice to work. In the VoIP industry, it's accepted that faxing over VoIP is never guaranteed-reliable. Sometimes it can be made to work, but it's easy to break. Fax machines use modems to communicate, just like your heart monitor.
   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...

this man knows what he's talking about ;)

In general, in my opinion, headsets that are created specifically for VoIP use are generally garbage, for the reasons /u/the_real_swk said 8khz mono (maybe 16khz). That is going to sound like crap compared to even the cheapest stereo usb headset when you are using anything but pstn calling. If all you do is PSTN calls and you want cell phone quality ... a VoIP headset is right for you, otherwise get a gaming headset.

I personally use the Logitech G930. It's super convenient to have wireless, and in general I would say it's a great value. I have a Sennheiser dw Pro 2 (a $350 wireless voip headset) and it sounds like trash compared to the G930's.

I am generally opposed to VOIP over wireless. However, in the past I have setup a lot of wifi voip phones from Snom, Grandstream and some "whitebox" manufacturers and the users have generally been happy with them.

So, as long as you are aware that I believe you will end up having call quality issues at some point. Easiest method is to Buy one of these. Second easiest is to find an old wireless router that supports DD-WRT, install that and set it to AP Client mode. Third is going to be going through the various Windows utilities or trying to setup Internet Connection Sharing on your Windows laptop.

Or (!), see if your company has a license for the Altigen Max Communicator which is a software IP phone you can run on your desktop. Set that up, plug in a pair of good headphones, and use that as your phone.

This is a pretty high-level book that I bought when I was first learning: https://www.amazon.com/Packet-Guide-Voice-over-administrators/dp/1449339670/ref=sr_1_1?s=books&ie=UTF8&qid=1491019238&sr=1-1&keywords=voip

It's cheap and fairly short.

This is a much longer, more expensive, but waaaay more comprehensive look at telephony as a whole and the movement to VoIP: https://www.amazon.com/Telecommunications-Crash-Course-Steven-Shepard/dp/0071832661/ref=sr_1_1?s=books&ie=UTF8&qid=1491019425&sr=1-1&keywords=telecommunications+crash+course

For you, I would suggest the first one.

And even though the internet has turned on him recently (he did get pretty weird), Eli the Computer Guy has a good high-level video on it: https://www.youtube.com/watch?v=2x3Ie6VZ_sg

So looking at the various models we are considering for a deployment, it looks like each phone is going to utilize 6.3 watts of power.

• CP-7910G (6.3W)
  
• CP-7912G (6.3W)
  
• CP-7940G (6.3W)
  
• CP-7960G (6.3W)
  
  SOURCE: Cisco POE Power Requirements Page
  
  Does this mean if I wanted to have 8 phones active, I should have 50.4 watts in the power budget, or should I have more?
  
  This Netgear Switch seems like it would be great with 70 watts of power budget available. Right?

I have used these to great success in the past with asterisk-based systems. They make an 8-port version too.

However, it's always better to use a SIP trunk instead, with proper QoS on your firewall.

Best book on the SIP protocol
https://www.amazon.com/SIP-Understanding-Initiation-Protocol-Telecommunications/dp/1607839954

Best book on SIP Trunking
https://www.amazon.com/gp/aw/d/B003ATPQEI/ref=dbs_a_w_dp_b003atpqei

My first question is are the phones turning on? If the phones turn on when plugged into the wall, but doesn't when plugged into the switch, the phones are getting power from the core switch.

If this is the case, you're going to need a power injector or a POE(Power over Ethernet) Switch to power the phones. In a previous company we used this one with some good luck.

Agreed! WiFi phones are an expensive novelty. A DECT 6.0 phone connected to an ATA works best. If you need more range they make DECT 6.0 repeaters. DECT 6.0 operates at 1.9 GHz. This is a dedicated frequency space for DECT 6.0 phones ONLY. No worrying about WiFi/Bluetooth interference.

Right. If you want to go with Cisco, you could get something like a VG224 or just get enough FXS cards for an existing Cisco router. You may also need a PVDM card in the router if you don't already have one.

Your other option would be to use just about any ATA device. Cisco makes the ATA187s which has 2 ports. There's a lot of 3rd party ATA devices you could use such as this 8-port Linksys one- http://www.amazon.com/Linksys-Cisco-Telephony-Gateway-Spa8000-G1/dp/B000V2PHZA

I'm a generalist who got hired into a communications company back in April to do networking and build some cloud crap. VOIP and Unified Communications got me up to speed pretty quickly. It's a pretty good history of telephony, overview of the protocols, and various services and devices. It won't prepare you to administer a system, but rather to start taking whatever vendors certs, and to understand what you're looking at in packet captures.

Ok, sorry it's taken me so long to respond--hopefully you're still willing to look at this.

I don't think we have a PBX system, if I'm understanding the term correctly. We have a Panasonic KX-TG4500 system that our standard phone lines just plug into. My understanding is that this won't work with a VOIP system?

We need ~7 handsets and a fax machine.

Thanks!

Consider the Panasonic KX-TGP550 SIP DECT.

http://www.amazon.com/Panasonic-KX-TGP550-SIP-DECT-Phone/dp/B002SUEQBY/ref=sr_1_3?s=office-products&ie=UTF8&qid=1451685963&sr=1-3&keywords=panasonic+sip

Supports up to six phones and eight lines. Can do everything you want except for record calls. I think there are other means available to record calls.

In that sort of an instance, something like this would help:

https://www.amazon.com/Bridge-Public-Announcement-System-Amplifier/dp/B0037TO4IK

It's as Mango123456 said. However, you could add a device like The Stick http://amzn.to/1RYLwZb which comes after your OBI and has a Fax port and a Phone port. It will listen for the fax tones on incoming calls and route the call to the appropriate port. If you turn the ringer off on your fax then you will not get a ring when a fax comes in.

https://www.amazon.com/Headset-Buddy-Smartphone-Adapter-PH35-RJ9a/dp/B004FV7MHE

Should do what you want. Most mid-high range SIP phones have a second RJ9 port.

Go with a DECT cordless SIP. Anything WiFi driven requires top notch coverage and is generally more expensive.

Not the greatest phone, but fits in a Grandstream setup (no offense) http://www.amazon.com/Siemens-Gigaset-Cordless-Landline-A580IP/dp/B002DEMELO