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Circuit wrote:
Mar 9th, 2018 5:46 pm
So the voip clients on android that support push are, acrobits groundwire, acrobits softphone and bria mobile? Has anyone been able to properly configure push with any of these clients and voip.ms? I'm trying with Bria mobile (using their subscription app) and I notice that when it registers with the Bria push servers (CounterPath Bria Push Server 2.0.0 17783) it shows "No registration found" sometimes on the voip.ms portal home page. When that happens I can't receive calls. I can make calls no problem.
I've been using Acrobit GroundWire for years with push on VoIP.ms.

It's been super reliable.

Only time push went down is because I changed phones, but Acrobit support was really helpful in resolving this.
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I'm evaluating Zoiper right now, as it was the most popular independent app that I've seen mentioned. I haven't gotten to use it much yet, though.

Apparently one of the less obvious factors for these apps is battery consumption when they are set for being able to receive calls. Zoiper Pro (paid version) has only increased my consumption about 10%, which is not the deal-breaker that I feared.
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IanBrantford wrote:
Mar 12th, 2018 12:02 pm
I'm evaluating Zoiper right now, as it was the most popular independent app that I've seen mentioned. I haven't gotten to use it much yet, though.

Apparently one of the less obvious factors for these apps is battery consumption when they are set for being able to receive calls. Zoiper Pro (paid version) has only increased my consumption about 10%, which is not the deal-breaker that I feared.
I was using Zoiper for a year. I found Zoiper would lose the registration at voip.ms too often. Finally tried out Acrobits Softphone which uses push notifications. ($10). No more missed calls. In my opinion it's worth it.
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marc_t wrote:
Mar 12th, 2018 10:47 am
I've been using Acrobit GroundWire for years with push on VoIP.ms.

It's been super reliable.

Only time push went down is because I changed phones, but Acrobit support was really helpful in resolving this.
Thank you Marc, I'll give it a shot then. I still get the occasional "No registration found" on Bria. Too many settings and no google finds for voip.ms + bria mobile push operability.

Would you have any ideas on the differences between groundwire and softphone (Both Acrobits) guessing groundwire is just more feature rich? Their website doesn't differentiate too much. Wish they had a feature table comparing the 2.
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Dukman wrote:
Mar 12th, 2018 7:57 pm
I was using Zoiper for a year. I found Zoiper would lose the registration at voip.ms too often. Finally tried out Acrobits Softphone which uses push notifications. ($10). No more missed calls. In my opinion it's worth it.
Thanks. While I haven't gotten to use it much yet, I did catch Zoiper having lost registration when bringing the phone from standby and wondering how long it had been lost. Acrobits was the other one that I had wanted to try, so perhaps I'll just to straight to it next.
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Circuit wrote:
Mar 13th, 2018 8:59 am
I still get the occasional "No registration found" on Bria. Too many settings and no google finds for voip.ms + bria mobile push operability.

Would you have any ideas on the differences between groundwire and softphone (Both Acrobits) guessing groundwire is just more feature rich? Their website doesn't differentiate too much. Wish they had a feature table comparing the 2.
Push for Bria requires paying for a subscription.

For differences between Softphone and Groundwire, visit
https://www.acrobits.net/hesk/knowledge ... category=9
Transfer and attended transfer are only available with Groundwire.

Call conferencing is also missing from Softphone.

https://www.acrobits.net/hesk/knowledge ... article=32

Note that using Push requires you to give your SIP credentials to a third party. However, Acrobits can give you the software to admin your own SIPIS server, where you live, if you're willing to put in the work and keep a server running 24/7: https://doc.acrobits.net/sipis/installation.html. Also, visit https://forum.acrobits.net/viewtopic.php?f=1&t=101#p391.
And, in that case, it's not all that secretive and spooky.
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I have tried both Zoiper Pro and Acrobits Groundwire for audio quality, talking to someone on a good landline. Calls did A/B testing using, at my end:
1. Landline
2. PC Mobile (Bell) cellular
3a. Zoiper Pro (on my callphone) and voip.ms on WiFi
3b. Zoiper Pro (on my callphone) and voip.ms on LTE
4a. Acrobits Groundwire (on my cellphone) and voip.ms on WiFi
4b. Acrobits Groundwire (on my cellphone) and voip.ms on LTE

All calls offered nearly indistinguishable incoming audio to me. However, outgoing voice quality was not. The receiver reported that #2 was best, followed by #1. However, #3 was poor and #4 was fair at best. 3 and 4 were intelligible, but I sounded far away. There is no way that anyone who is hearing-challenged or experiencing background noise could hear me.

I don't see any codec controls on these apps, and the only gain control is "Speaker Gain" in Zoiper Pro, which did not improve anything.

Any hints?
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May 19, 2003
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Bria will allow choosing codecs. But you can also choose codecs in voip.ms. Just only use G.711u and it'll be indistinguishable from landline.

IanBrantford wrote:
Mar 16th, 2018 2:52 pm
I have tried both Zoiper Pro and Acrobits Groundwire for audio quality, talking to someone on a good landline. Calls did A/B testing using, at my end:
1. Landline
2. PC Mobile (Bell) cellular
3a. Zoiper Pro (on my callphone) and voip.ms on WiFi
3b. Zoiper Pro (on my callphone) and voip.ms on LTE
4a. Acrobits Groundwire (on my cellphone) and voip.ms on WiFi
4b. Acrobits Groundwire (on my cellphone) and voip.ms on LTE

All calls offered nearly indistinguishable incoming audio to me. However, outgoing voice quality was not. The receiver reported that #2 was best, followed by #1. However, #3 was poor and #4 was fair at best. 3 and 4 were intelligible, but I sounded far away. There is no way that anyone who is hearing-challenged or experiencing background noise could hear me.

I don't see any codec controls on these apps, and the only gain control is "Speaker Gain" in Zoiper Pro, which did not improve anything.

Any hints?
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shutterbug wrote:
Mar 16th, 2018 7:23 pm
Bria will allow choosing codecs. But you can also choose codecs in voip.ms. Just only use G.711u and it'll be indistinguishable from landline.
I'll give that a try. It took a while, but I found the setting. Thanks.
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IanBrantford wrote:
Mar 16th, 2018 2:52 pm
I have tried both Zoiper Pro and Acrobits Groundwire for audio quality, talking to someone on a good landline. Calls did A/B testing using, at my end:
Provided an app isn't buggy, SIP apps should have minimal affect on incoming sound quality when using the same provider and codec.
However, outgoing voice quality was not.
You're using different handsets/mics, so it's not as though the same hardware is being used in all tests.

Groundwire sound settings can be found under Settings-->Preferences-->Sound
If you increase Microphone volume boost, you may find you need to disable noise suppression. Sometimes those two settings don't work well together.


Also, it should be noted that available upload bandwidth can affect outbound sound quality. Choppy audio/jitter can be ameliorated by properly configuring QoS in routers (matters when on Wi-Fi) for your VoIP device.
For a general QoS description, visit http://www.voipmechanic.com/qos-for-voip.htm

Using cellular data for VoIP can a bad idea depending on signal strength. The two biggest problems for VoIP are jitter (large variation between each successive ping) and pings (delay). Jitter affects how choppy a call can sound (if jitter/packet loss is horrible enough, a call can drop). The larger the pings, the greater the delay from the time someone speaks to the time it's heard. That's why it's a good idea to use SIP servers that are close to you.


I don't see any codec controls on these apps
All of them should allow you to select codec priority. Groundwire certainly does under Accounts-->Provider-->press "i"-->Advanced settings
Then you can select priority for codecs for Wi-FI and codecs for Mobile data.

G.711u obviously sounds better than G.729a (at the expense of more data being used).

All of my calls on Wi-Fi sound fantastic, but we may not be using the same hardware, and I have QoS properly configured in my router for all of my VoIP devices.

With VoIP.ms, sometimes the routing (Premium vs. Value) used can affect call quality to a specific destination since different carriers are used. If a carrier is at fault, fiddling with app settings is going to accomplish nothing.
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Webslinger wrote:
Mar 18th, 2018 1:43 pm
Provided an app isn't buggy, SIP apps should have minimal affect on incoming sound quality when using the same provider and codec.
Good. That was my experience. Incoming audio was very good.
You're using different handsets/mics, so it's not as though the same hardware is being used in all tests.
??? I was using the same handset for all tests. I just tried different diallers (Android default, Zoiper Pro, or Groundwire). I neglected to try the default Android dialler's SIP support. It would be interesting to know whether it gives the same oddly consistent problem (low outgoing volume) as Zoiper and Groundwire.
Groundwire sound settings can be found under Settings-->Preferences-->Sound
How did I miss this when I went looking for it? I even googled for it and found nothing, yet there it is. Thanks. It's exactly what I wanted. Wow, there are several other interesting things in there as well, such as "remember audio route" (which is off by default).
If you increase Microphone volume boost, you may find you need to disable noise suppression. Sometimes those two settings don't work well together.
Thanks again. I'll mind this and possibly try variants of these settings.
Also, it should be noted that available upload bandwidth can affect outbound sound quality. Choppy audio/jitter can be ameliorated by properly configuring QoS in routers (matters when on Wi-Fi) for your VoIP device.
For a general QoS description, visit http://www.voipmechanic.com/qos-for-voip.htm
This reminds me that I recently switched to an ADSL2 Internet service that seems to wallow in self-pity under load, meaning that only one large task can happen at once -- everything else slows to a crawl. I might upgrade back to FTTN to fix this and get more consistent service. The price difference isn't that much.
Using cellular data for VoIP can a bad idea depending on signal strength. The two biggest problems for VoIP are jitter (large variation between each successive ping) and pings (delay). Jitter affects how choppy a call can sound (if jitter/packet loss is horrible enough, a call can drop). The larger the pings, the greater the delay from the time someone speaks to the time it's heard. That's why it's a good idea to use SIP servers that are close to you.
It will have to work reasonably well on LTE and just passably on HSPA, as this experiment is to determine whether a VOIP system using cellular data can let me and several family/friends get by using that for calls in lieu of a voice plan. Many people seem happy with it on LTE, so it's a big savings on subscriptions that don't require a voice plan. Ottawa is between major servers in Toronto and Montreal. We'll see.

All of them should allow you to select codec priority. Groundwire certainly does under Accounts-->Provider-->press "i"-->Advanced settings
Then you can select priority for codecs for Wi-FI and codecs for Mobile data.

G.711u obviously sounds better than G.729a (at the expense of more data being used).
I found that in Groundwire, but will test all three SIP options on my phone further using server settings as recommended earlier, using just one codec at a time.
All of my calls on Wi-Fi sound fantastic, but we may not be using the same hardware, and I have QoS properly configured in my router for all of my VoIP devices.
I'm happy to do some jiggery-pokery with the router. Most important is mobile though, and I'm even willing to just stay off of WiFi if the router won't accommodate.

With VoIP.ms, sometimes the routing (Premium vs. Value) used can affect call quality to a specific destination since different carriers are used. If a carrier is at fault, fiddling with app settings is going to accomplish nothing.
Oohh. I'm on the Premium right now. Though I can see call quality suffering irregularly in ways that introduce distortion or drop-outs, I'm not sure how volume would be uniformly affected. Anything is worth trying if the obvious things to try (microphone gain) don't work. Thanks again.
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IanBrantford wrote:
Mar 18th, 2018 9:31 pm
??? I was using the same handset for all tests.
IanBranfford wrote: 1. Landline
A landline typically isn't a smartphone, unless you're using bluetooth to connect to a wireless handset. Possibly, I'm misunderstanding something.
Also, it wasn't clear to me that the same smartphone was used with PC Mobile (#2 in your post).
This reminds me that I recently switched to an ADSL2 Internet service that seems to wallow in self-pity under load, meaning that only one large task can happen at once -- everything else slows to a crawl. I might upgrade back to FTTN to fix this and get more consistent service. The price difference isn't that much.
The concern would be that other devices that are connected to your LAN are hogging upload bandwidth without your knowledge (upload for outgoing audio). QoS helps to ensure there's sufficient bandwidth available for your VoIP devices (in this case, your smartphone) at all times. What QoS won't fix is unreliable/poor internet service.
It will have to work reasonably well on LTE and just passably on HSPA, as this experiment is to determine whether a VOIP system using cellular data can let me and several family/friends get by using that for calls in lieu of a voice plan. Many people seem happy with it on LTE
The first step of using VoIP is to have reliable internet connectivity. Cellular data, universally, doesn't provide that stability. And Wi-Fi, at home, may be better for some, but it's still not as reliable as a wired Ethernet connection.

So, to me, someone asking me what cellular service to use for VoIP (and I'm starting to be asked this a fair amount in person) is like asking what's the best way to run across a partially frozen lake in the winter. Probably if I stay in certain areas, I'll be reasonably okay, but I'll never be close to being 100% secure. And then if the wind blows a little (or a lot if someone is actually using Wind . . . errr Freedom Mobile), or I accidentally end up in an area where the ice is weak, I'll fall through.

With Robellus, it's impossible to remain on LTE constantly while traveling on highways between towns in Ontario, for example.

Here are some basic concepts involving jitter and ping (I've written this before for general advice; I'm just copying and pasting my own writing) . . .

People should be testing their pings and jitter (you want little to no variation between pings) to the specific VoIP providers' SIP servers they plan on using before purchasing anything.

My pings to

a) voip.freephoneline.ca average 11 ms.
b) voip2.freephoneline.ca average 12 ms
c) voip4.freephoneline.ca average 27 ms

I also see low latency to voip.ms closest sip servers to me as well.
http://wiki.voip.ms/article/Choosing_Server


My pings to VoIP sip servers (FPL, Anveo, voip.ms and to ping.callcentric.com), are well below 50.

Anything over 200ms is unacceptable. You'll begin to encounter crosstalk, even if an untrained ear doesn't notice. So, if you're getting really high pings and jitter, I would try to do something else (choose a different service provider that's closer to me; enable QoS in my router if I'm on Wi-Fi or ethernet; ensure I'm in a good LTE area; etc.).

What you don't want to see is 40, 45, 50, 35, 500, 40, 30, 45, 700. That's bad jitter.
You want relatively consistent pings without a lot of variation.


Ooma is selling a proprietary device and a single service (Ooma's).
Their SIP servers are located in California. They also a SIP server location on the U.S. East Coast now, but I haven't
looked into where that location on the East Coast is. I'm not positive
whether they will offer something more local for Canadian customers in the future.
http://www.monitis.com/traceroute/
208.83.244.94 is one Ooma SIP server that seems to be on the west coast of the U.S.
Regardless, Ooma is a U.S. company, based in California.

Some popular voip services include
freephoneline.ca (servers are in Ontario, I think, possibly around Waterloo, but I'm not positive), voip.ms (wide range of server locations: http://wiki.voip.ms/article/Choosing_Server), anveo.com (Montreal), www.thespout.ca (Vancouver and Seattle), and callcentric.com (New York).




1.Use winmtr http://winmtr.net/download-winmtr/. Ping about 100 times.
When using WINMTR, look at the very last line or hop when checking your pings.

If you're on a Macintosh, maybe this helps: https://www.reddit.com/r/TagPro/comment ... tr_on_mac/

When using WinMTR, look at your average ping and then maximum ping. Although WINMTR doesn't provide a jitter value, you can get an idea of what yours is by subtracting maximum ping from your average.
Jitter is the difference between each successive ping.
The bigger the difference, the bigger the problem.

Same with ping, which represents lag or delay. The lower your ping and jitter, the better.

Generally speaking it's best to have a decent router for VoIP with strong QoS features.
Stick your ISP's modem/router combo in bridge mode, use your own router, and properly enable QoS in your router for your ATA.



2. For Freephoneline.ca (Ontario based, possibly around Milton or Waterloo), test to voip.freephoneline.ca (let winmtr ping about 100 times), voip2.freephoneline.ca, and voip4.freephoneline.ca. You can copy text to clipboard and paste your results (do not post your own IP public address though) and post them for others to examine if you want.

You do not want high pings and lots of jitter (you do not want a lot of variation between each ping). If you get horrible results, you should probably avoid FPL.

You should also test to make sure FPL works for you before paying anything: https://www.fongo.com/app/desktop/. You will need a microphone and speakers (or preferably a headset).

3. For voip.ms, test to the closest server to you:
http://wiki.voip.ms/article/Choosing_Server

This company has a lot of servers in a lot of different locations.

4. For Anveo (Montreal POP), test to sip.ca.anveo.com

POP= Point of Presence
That's essentially an access point or physical location where two or more types of communication or network devices make a connection.

5. For The Spout (Vancouver), test to ca.sipfrom.thespout.ca
Spout Communications also has servers in Seattle.

Spout Communications can obtain phone numbers for a lot of rural areas in Canada.

Update . . . There may be some issues with Spout: http://www.dslreports.com/forum/r311931 ... ne-call-Or.

6. For Callcentric (New York), test to ping.callcentric.com

7. For Ooma (California), test to myxprov.ooma.com
If I find the east coast server address (or if someone PMs it to me), I'll post it.

By the way, with WinMTR, you should also test to 74.125.39.7 (California) if you plan on trying to use Google Voice.
Obtaining a Google Voice number requires that you have a U.S. IP address and a U.S. phone number first (and I will generally be avoiding questions on how to go about obtaining a GV number).


Pinging and testing to the SIP servers you're thinking of using (and to Google Voice) should always be the first steps before jumping into a voip service.
Do this between 8 p.m. and 11 p.m. to see if local congestion is a factor (this often is your ISP's fault). Sundays are the best days to test (because that's when most people in your area will be home). 8 p.m. - 11 p.m. is prime time.

If you get horrible results (really high pings and jitter), do not sign up for the service. You will not be happy.
This goes for all VoIP services. Test to their servers first.

This is one reason why I suggest testing first . . .

Ping is a measurement of data packet transmission, and ping does affect delay or lag. All gamers know, almost inherently, that lag affects them negatively. A PC gamer will pound his or her keyboard in hope that a character will respond on his or her monitor, quickly, but when there's a delay or lag, reality doesn't meet expectation. A gamer can see this problem visually. Over VoIP, anything over 200-210 ms, you will typically start to encounter crosstalk due to increased delay, even if the untrained ear doesn't notice. All VoIP services are subject to the same scientific principles including the fact that speed of transmission affects delay, and Ooma is not some magical service that is somehow exempt from issues arising from high pings and jitter. I have helped a few people with Ooma. When pings (and especially) jitter are high, it's a pretty horrible experience, just as it would be with any other VoIP service. When pings and jitter are fine, so is Ooma.

Paul's not having jitter issues, but he is experiencing delay:



Start at the 48 minute 20s mark.

https://tinkertry.com/why-i-gave-up-on- ... ne-service
Paul wrote:
"I figured it was time for one last email to Ooma. Can they give me a way to connect my phone calls from a server a lot closer to Connecticut than San Jose, California? It got me a response the next day that gave me a bit of a chuckle


Dear PAUL,

Thank you for contacting Ooma Customer Care. Good day! We are sorry that this isn`t going to work for you. As mentioned before, we only have one server which is in west coast as of yet and we do not have control over with this latency.

If you decide to port your nos. out of Ooma, we will need to keep your account active while the other provider is in the process of porting your nos. out. Please let us know once that is completed so we can remove your nos. from our database.

In case you want to stay with us, we can refund half of the amt. you paid for the Annual Premier.

Please write me back if you have further questions and I will respond to you as quickly as possible.

Thank you for choosing Ooma!

Sincerely,

Ooma Customer Care Specialist
Chat Support is now available 24/7
To reach a live chat agent, please visit us at www.ooma.com/support"

Ooma now has an East Coast server. So Paul may want to test with Ooma again. Back when he was using Ooma, the only server location was in California. However, Ooma may be using the cheapest routing options available after its venture capital ran dry (according to some people), which may make some call quality in some cases less than ideal: http://www.dslreports.com/forum/r30911315-

So go ahead and test.

Anyone using any communication service (or even when playing online games or using other online services) should understand that the longer the path to the server being used, the greater the potential exists for a problem to occur somewhere along that path. This is one reason why voip.ms is so desirable to some people; it has SIP servers situated in numerous locations (not just in California).


Another factor to keep in mind, is that during prime time (between 8 p.m. and 11 p.m.) cable internet nodes may be oversubscribed and face congestion issues (and congestion can also exist with DSL). So I suggest testing services between 8 p.m. and 11 p.m., particularly on Sundays, when everyone in your area will be home.

Running http://vac.visualware.com/index.html at 8p.m. (especially on Sunday) may be a better test than a speedtest. Or pick the server that's closest to your VoIP provider's server. A MOS score below 4.0 is bad news. It means call quality will not be good.



IanBrantford wrote: I just tried different diallers (Android default, Zoiper Pro, or Groundwire). I neglected to try the default Android dialler's SIP support. It would be interesting to know whether it gives the same oddly consistent problem (low outgoing volume) as Zoiper and Groundwire.
For what it's worth, jitter (choppy audio) and ping (delay) are not likely to affect outbound volume. Mics in handsets might (a different smartphone may provide different results). Mic gain should. Some bug/compatibility issue between a SIP app and a smartphone's firmware might. You might want to google "Nexus 6P low mic voip" for some general info for what can happen with a smartphone. Occasionally, a carrier might.

Oohh. I'm on the Premium right now.
Oddly, in a few instances Value routing was better for me when making calls to a rural area in B.C. using VoIP.ms, until I contacted them to complain. Then they changed the carrier for me pretty quickly after I wrote,
"Well, I'm certainly happier paying less with Value rates if you're not going to switch carriers." They've also changed routing for me in the past when I was having issues calling Japan. It depends on the carrier being used, and there can be a difference sometimes between Value and Premium (and sometimes the difference isn't obvious, given the example in B.C. that I mentioned).

Anyway, good luck
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.55) can be found here. OBi200 info can be found here. For OBi202 info, click here.
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Does VOIP allow verification codes through? Like for 2FA. I've tried having verification codes sent to burner numbers via apps like TextNow but they never worked... maybe it's a limitation of the app, IDK.
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cellnerd wrote:
Mar 19th, 2018 6:38 pm
Does VOIP allow verification codes through? Like for 2FA. I've tried having verification codes sent to burner numbers via apps like TextNow but they never worked...
Try using the burner # from TextmeUp. I have verified Whatsapp # that way. TextNow just couldn't do it.
That does not cost you a penny
Daniel

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Webslinger wrote:
Mar 18th, 2018 10:29 pm
A landline typically isn't a smartphone, unless you're using bluetooth to connect to a wireless handset. Possibly, I'm misunderstanding something.
Also, it wasn't clear to me that the same smartphone was used with PC Mobile (#2 in your post).
Oh, you're not misunderstanding. I can now see that I had a repeated typo... what's a "callphone"? My cellphone, that's what. I wasn't using an ATA device or anything like that. For those counting, that's three mistakes that I made in one post. Anyway, my cellphone was the common equipment for all VOIP tests (cases 2-4). Sorry about that.

Current status: adjusting the microphone gain in Groundwire fixed the problem very well, and now that my golden-eared test subject can hear me loudly enough, the report is that quality was quite good. I'll do more tests on LTE to verify. I only need "fairly good" audio quality, as most phone calls aren't of a nature that requires the best. Delayed speech as in your video example would be a moderate problem. I'm currently getting a ping time of 12 ms on wired home Internet, 40 on WiFi, and 60 on LTE. It's pretty consistent at my home.

The main outcome now is that it's definitely worth porting my primary mobile number to voip.ms, as I get the benefit of ring groups, so I can send incoming calls to all of my phones (1 landline and several mobiles) without forwarding or swapping SIM cards. This nearly 20-year-long first-world problem can go away.

The only things that I have to determine for myself are:
a) reliability on LTE around town
b) Groundwire's interactions with my Pebble smartwatch

Groundwire is sending frequent Push notifications to my watch (unwanted and annoying) but NOT incoming calls. If I can't do the right jiggery-pokery to fix that, I'll settle for a conventional voice plan and send incoming calls to that, using Groundwire only for outgoing calls on my primary phone.

As long as reliability is good after a couple of weeks of testing, I'll be recommending a VOIP solution with Fido's 3GB tablet plan to some friends and family, if we can get them. If we can't get them independently via WirelessWave, I'll conscript some friends with Fido accounts to procure them. :-)
The concern would be that other devices that are connected to your LAN are hogging upload bandwidth without your knowledge (upload for outgoing audio). QoS helps to ensure there's sufficient bandwidth available for your VoIP devices (in this case, your smartphone) at all times. What QoS won't fix is unreliable/poor internet service.
I'm willing to delve into that, but it occurs to me that, since I'll be using voip ring groups to send incoming calls to my landline, I won't need to bother at this time. I'll revisit for other people or if I give up my landline in future.
The first step of using VoIP is to have reliable internet connectivity. Cellular data, universally, doesn't provide that stability. And Wi-Fi, at home, may be better for some, but it's still not as reliable as a wired Ethernet connection.
I just need it to be fairly good on mobile, and reliable -- not quite as good as conventional cellular calls. I use a landline at home for important planned calls, and now that I no longer call in to conference calls from home... that's hardly ever.

Oddly, my mom, er, my test subject reported that my cellular call on Bell was even higher quality than the (3rd-party) landline. Once volume was corrected, WiFi-based VOIP calls on the cellphone were between those two in quality, i.e. everything was quite good. We'll see about LTE in the urban areas.
So, to me, someone asking me what cellular service to use for VoIP (and I'm starting to be asked this a fair amount in person) is like asking what's the best way to run across a partially frozen lake in the winter. Probably if I stay in certain areas, I'll be reasonably okay, but I'll never be close to being 100% secure.
This is why I would not want a mobile service that completely rules out regular voice calls, even at higher cost. I just want VOIP for almost all usage.

Another bonus of finding those low-level settings in Groundwire: it has a programmable special call routing feature. For example, you can route 911 emergency calls through the conventional cellular voice service for possibly higher voice quality and location service, even when you instinctively use the Groundwire dialler.
And then if the wind blows a little (or a lot if someone is actually using Wind . . . errr Freedom Mobile), or I accidentally end up in an area where the ice is weak, I'll fall through.
Haha! Point taken. One person for whom I might recommend this kind of switch is currently on Freedom, but would be happy to move to Fido 3GB tablet plan if voip telephony works well enough.
With Robellus, it's impossible to remain on LTE constantly while traveling on highways between towns in Ontario, for example.
I'm often in those areas for day trips, but very seldom making calls there. So, it's not a deal-breaker. It would be if I lived there.
Here are some basic concepts involving jitter and ping
(Awesome brain dump snipped)
Anyway, good luck
Thanks!

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