VoIP Box Review & Setup Guide (SPA-2102)

[js]

In the interest of future readers and basic stuff, one thing that might be helpful to mention is about local IP addresses.

If you can't seem to connect to your VoIP box or router, then change the IP address of your computer such that the first three parts of the IP address are the same, and the last one different. This is especially a pro pos if you change the IP address of your VoIP box or router. For example, my 2102 came loaded with a static IP of 192.168.0.1, but my router was 192.168.1.1, and was handing out dynamic IP addresses starting at 192.168.1.100. So, I wanted to change the VoIP to 192.168.1.1 and the router to 192.168.1.2. As soon as I changed the IP of the VoIP, I lost connection with it, and even after typing in http://192.168.1.1 (the router was off-line, powered down, and already changed to 192.168.1.2 in any event), I got nothing. I immediately knew why and went into my network settings and set the IP of the laptop to 192.168.1.107, and then reloaded the URL of the VoIP box GUI, and voila, I had access again. If your subnet mask is 255.255.255.0, it means that the first three parts of the IP address of all the things on the LAN had better be the same, otherwise, they ain't all on the same LAN and won't talk to a web GUI that requires you be on the same LAN.

Oh, and that also means that if your VoIP or router or whatever is a DHCP server, that you probably want it to be giving out local IP addresses that have the same first three numbers as itself. i.e. before I changed the 2102, it's IP was 192.168.0.1, and it was giving out IP addresses starting at 192.168.0.2. I changed its IP to 192.168.1.1 and had it hand out IP addresses starting at 192.168.1.100. I had to make both changes. Think of the first three numbers as putting all the stuff on the LAN in the same room, and the fourth number giving them ID numbers so they can recognize and talk to each other. If you give two people the same number, it's bad, as they both answer to the same ID. Bad!

Please correct anything that isn't quite right here, ElectronGuru. Thanks!

[ElectronGuru]

And . . . I did indeed buy an Airport Extreme, with the simultaneous dual band feature. I had other stuff to do yesterday evening and today, but I did start digging into Apple's 77 page online manual, and just like I thought, this thing can do all sorts of sophisticated stuff, including WDS (wireless bridging, if you have two, or one of these and an airport express, etc.), but at the same time it is super easy to do the easy stuff, and probably pretty easy to do the difficult stuff, too. Probably all thanks to the Airport Utility program built into OS X . . . but still.

Airport Utility is available for Windows too.

I consider the specialness (hardware+software) of airport base stations to be almost on par with iPods. Mine has a built in drive, on which sits about 800GB of music, movies, and TV shows, drawn from computers throughout the house. With the right settings, you can wirelessly stream 1080P! Speaking of which, assuming its still optional, turn on Use Wide Channels. Located under Airport>Wireless>WirelessOptions, it doubles the capacity of your N connections.

I'll report more later, once I've got everything up and running, but I did use the 2102 VoIP box to call my brother and talk for a bit, and I was very impressed with the sound quality. Pretty much as good as a plain old telephone line.

Nice.

I figured getting the VoIP box up and running was going to be the tricky part. Go figure. Now I understand why I've heard some people criticizing DD-WRT, in any case!

For me that was the hard part. I hope this guide contributed a bit.

In the interest of future readers and basic stuff, one thing that might be helpful to mention is about local IP addresses.

Please correct anything that isn't quite right here, ElectronGuru.

Great advice (and correct) I would only add two extra bits.

DHCP works via a lease system. A dhcp server is configured with a set time per lease, usually a few hours. When a given machine's lease is up, it asks for a new lease - and generally gets the same address. If you're playing around with a network (as we are doing here) and address schemes are changed, it may be handy to get client machines to ask for a new lease sooner - getting them on the new scheme immediately. This is done in a variety of ways, depending on the OS. Recent version of OS X have a button in the Network preferences called Renew DHCP Lease. Press it, and a few second later, the new IP shows up and the network is ready to use.

In the midst of these changes, you can also change the private IP range in question. There are actually 3 private ranges to choose from:

10.0.0.0 – 10.255.255.255
172.16.0.0 – 172.31.255.255
192.168.0.0 – 192.168.255.255

Any private range can be any of these or any subset of these. Somehow, the smallest and most awkward (IMHO) became the standard. I like to set mine to 10.0.0.1+

[js]

Thanks for the info, ElectronGuru, and yes, this guide was, I'm positive, the main reason for the VoIP setup being easy. I really appreciate it! I certainly wouldn't have done the VoIP thing without it.

But, seriously, DD-WRT is screwed up. It takes about 2 minutes (tops) to set an Airport Extreme up in bridge mode. You tell it to get a DHCP addy from the WAN port / DHCP server. And you tell it to turn off (bridge mode) IP sharing. That's it. TWO SETTINGS (approximately). Two. Easy. Peasy.

Yet the thing can be set up for a roaming network with other stations, or can do wireless bridging, NAS, music streaming, network attached printer, etc. Not to mention the dual feature.

And yes, it does still have the wide channels option. Cool.

[ElectronGuru]

Hay, did you find the part of the AEB yet where you can plug an USB drive (including thumb drives) into it for use as a wireless file server!?

[js]

Hay, did you find the part of the AEB yet where you can plug an USB drive (including thumb drives) into it for use as a wireless file server!?

Yup.

Setting up the AEB was an absolute joy--the quintessential apple "experience", where stuff just works and flows and makes perfect sense and where one step just leads naturally into the next. Beautiful. Such a pleasure compared to effing DD-WRT. And from what I can see, there's pretty much nothing that DD-WRT can do that Airport Utility can't. Bridging the AEB was as super-straightforward as such a super-straightforward and common thing SHOULD be. I just put it in bridge mode = DHCP server off. TA-DA! Done. Worked.

It just worked. Lovely.

Anyway, moving on, I still need to update my firmware so we're on the same page here, but right now, I'm having a few small sound quality issues. It seems that when the person talking to me peaks their volume, that I get a cut-out of the sound. As long as they are talking quietly, this never happens. But if they raise their voice a lot at points, those points get not just clipped, but totally cut out, and I miss what they said.

I'm using the 711u codex as my preferred codex, but I don't have the option to "ONLY use preferred codex" set. And I notice that there are some silence supp settings I could change, and I suppose I could change the QoS setting, right? At the moment I have it at 80 percent of my bandwidth. I assume changing that to 70 percent might improve sound quality? Or would it?

Like I say, first thing I'll do is update my firmware and see where that leaves me. And even if the sound quality never gets better, I'm still going to totally switch over to this VoIP. The value is just too good to pass up. So a few words get clipped now and then--still better than a cell phone--and I'll be saving like $50 per month.

[ElectronGuru]

I still need to update my firmware so we're on the same page here, but right now, I'm having a few small sound quality issues. It seems that when the person talking to me peaks their volume, that I get a cut-out of the sound. As long as they are talking quietly, this never happens. But if they raise their voice a lot at points, those points get not just clipped, but totally cut out, and I miss what they said.

Have a look at these two settings under Voice>Region (Admin Login/Advanced)

FXS Port Input Gain:
FXS Port Output Gain:

These are like volume controls. If the problem is a true clipping from the sound being to loud, putting a larger negative number in here (-3 -> -6) may correct it. The same change also reduces echos

I'm using the 711u codex as my preferred codex, but I don't have the option to "ONLY use preferred codex" set.

Actually, I have the option and leave it OFF. Think of it like a fallback mode. If the first codec doesn't work for any reason, it can try the second. In this case, since the first (711u) uses more bandwidth this can be very handy when bandwidth is tight.

And I notice that there are some silence supp settings I could change, and I suppose I could change the QoS setting, right? At the moment I have it at 80 percent of my bandwidth. I assume changing that to 70 percent might improve sound quality? Or would it?

You want to be very conservative when it comes to settings. Change as few as possible, change as few at a time as possible, and change them as little/gradually as possible. Once you're down the rabbit hole (think Matrix), its a challenge to get back on solid ground.

That being said, there is no harm in writing down a setting you want to play with and changing it back when you are done. The firmware MAY be causing your sound issue, the QoS MAY be causing it, or the percentage MAY be causing it. The only way to know for sure is to try. But of these, the only likely suspect is the firmware.

As with preferred codec's, QoS is mearly a prioritization scheme. The best way to test if bandwidth is a factor with an issue is to unplug the rest of the network. QoS or no QoS, when the box is by itself - your voice packets are getting complete priority.

Like I say, first thing I'll do is update my firmware and see where that leaves me. And even if the sound quality never gets better, I'm still going to totally switch over to this VoIP. The value is just too good to pass up. So a few words get clipped now and then--still better than a cell phone--and I'll be saving like $50 per month.

Looking at my last few months, my main/voice line has 1000 minutes use, costing $6.38 (+monthly's). My fax line is $0.08 total. The value is silly better.

The firmware process on Mac's is a bit tricker because there is no dedicated firmware app. Let me know if you get hung up.

[js]

Well, damn it, I inadvertently locked myself out of access to the 2102--at least as admin.

I disabled web interface or something like that, because I thought it was asking me if I wanted to be able to modify the settings of the 2102 from the WAN. I can log in as user, but not as admin. Even if I am directly hooked to the ethernet port of the 2102, I can't log in as admin.

Is there anything I can do other than totally reset everything but the firmware?

[js]

Oh, and I did successfully update the firmware by uploading the .bin file to my webspace and putting the URL in the upgrade rule filed and restarting the box. I set my second preferred codex to 729a and set the QoS to "On when phone in use". I also undid the QoS Qdisc (whatever that is) from TBF to none, which is how it started.

Unfortunately, I changed the DMTF method to AVT, as they talked about that in the firmware upgrade notes, with the strict mode of holdoff and a 70 for holdoff time or something like that, but that was only for IF you were using AVT, and you told us to use Auto. I figured I'd change it in a second. Then locked myself out of admin access. Damn, damn, damn.

[ElectronGuru]

Nice work on the firmware. Just make sure to blank out the upgrade rule field when you're done or it will be flashing every time you restart.


So you changed the password for admin or just WAN access for admin? If its the later, try this:

pick up a phone, dial ****, then
7932# [plus password# if set], then press 1#


I've also experienced an issue where I could not log in as admin unless I first logged in as user. Try typing in the user name/pass on the first go, then clicking the admin link followed by admin name/pass.

Take a deep breath and remember: You can always reset the box (section D, last line), then follow from step 1 again. Think of it like unscheduled practice! Its also a good way to guarantee that all settings are as expected. Just make sure you have your Call Centric info first.

[js]

I disabled web access for admin. I can log in as user, but there simply is no admin link at all to click!

As for the **** and 7932#, that's all well and good, except that I do have a password entered for both admin and user, and they are NOT numbers? Is the "password#" you're talking about different than the admin password? Anyway, I'll try it and see what happens.

Although . . . I don't really need to, as I already blanked the upgrade rule field and updated (seemed like the smart thing to do), and more importantly,

. . . the sound quality is now . . .

EXCELLENT!

It could be the firmware upgrade, or it could be that I set DMTF to "AVT" and strict holdoff and a value of 70, just as the firmware notes suggested (if using AVT, anyway).

Is there a problem with using "AVT" mode as opposed to "Auto"?

Anyway, I could just leave well enough alone, really.

My next step is to port my old number over to Callcentric. I'll look into that tonight maybe. Or I may give the VoIP setup just a day or two more just to be sure.

Oh, and for anyone who decides to do this, I can give you the URL to my webspace .bin file for upgrading to the latest firmware version.

It's funny, in my phone bill for this month I got a notification of rate increases. HAH! Not for me, you bastards!

[js]

I did the **** thing, typed in my password using the letters to numbers translation on the keypad. It worked. I enabled WAN access. But I still can't log in as admin. Only as user.

[ElectronGuru]

Good stuff JS, we keep finding new commands! Try this:

723646# [plus password# if set], then press 1# (enable admin access)

more importantly,

. . . the sound quality is now . . .

EXCELLENT!

It could be the firmware upgrade, or it could be that I set DMTF to "AVT" and strict holdoff and a value of 70, just as the firmware notes suggested (if using AVT, anyway).

Is there a problem with using "AVT" mode as opposed to "Auto"?

Here's what the manual has to say:

DTMF Tx Method Select the method to transmit DTMF signals to the far end:
InBand, AVT, INFO, Auto, InBand+INFO, or AVT+INFO.
InBand sends DTMF using the audio path. AVT sends DTMF
as AVT events. INFO uses the SIP INFO method. Auto uses
InBand or AVT based on the outcome of codec negotiation.
The default is Auto.

Basically, Auto is more versatile. Change it back and forth and see if you can hear a difference.

It's funny, in my phone bill for this month I got a notification of rate increases. HAH! Not for me, you bastards!

Good feeling

[js]

ElectronGuru,

On the same page that you type in a user and admin password, there is an option that says something about "enable web access".

That's what I turned off.

And I can not effing unlock it.

And I'm worried that it won't unlock even when I do a factory reset. I'm worried I did what a provider does when they want to provide a unit to their customers that is locked to their provider settings.

Can you take a look at that page and see what you think? (but do not do what I did--do not disable that setting).

Should I call Linksys customer service? I wouldn't mind doing a hard reset and going through all the setup again, but I WOULD certainly mind doing a hard reset and bricking my 2102. If you google SPA2102 and "unable login admin" you'll read some horror stories.

Should I worry? Or what do you think?

[js]

Good stuff JS, we keep finding new commands! Try this:

723646# [plus password# if set], then press 1# (enable admin access)

OMG!

YES! YES! IT WORKED! Oh, thank you! Thank you! Thank you!

You don't know how much of a relief this is! Excellent! Really great!

You are THE MAN, ElectronGuru!

[ElectronGuru]

[ElectronGuru]

BTW, when you're away from home, you can configure your Call Centric account to forward calls to your cell or put SIP software on your MacBook and send/receive VoIP from any hotspot.

I've not tried it, but this SIP software is free (mac only)
http://code.google.com/p/telephone/

[cjames]

For the record, this is my configuration:

CABLE

- cable modem

WAN

- Apple Time Capsule (video server, nat/dhcp on)

LAN

- 2102 (nat/dhcp off)

Hi there,

Glad to hear that js got things going well.

I'm trying to set things up exactly how ElectronGuru has above. I got the SPA-2102 in the mail this morning and haven't had a chance to play yet, but the instructions i have from my ISP tell me to plug the SPA-2102 directly into my ADSL modem (and fill in the PPPoE fields etc).

I was wondering what settings I should fill in to have it setup as you do above. As things are now I have:


ADSL Modem (192.168.0.1)

(connected to)

Apple Time Capsule (nat/dhcp on; serves 10.0.0.x addresses to two laptops)



I assume I plug an ethernet cable into the lan port of the timecapsule and in to the wan port of the spa-2102. I assume that I don't have to worry about the PPPoE settings on the spa-2102 and that I'll need to set wan to DHCP and make sure that the DHCP server on the spa-2102 is disabled.

Are there any other settings i shoudl know about? Do I need to worry about setting QoS on my router or forwarding ports on the router and/or timcapsule?

Anyway I'll have a play tonight or tomorrow but finding someone knowledgeabkle who has the same setup is great. Any hints would be much appreciated!

[js]

ElectronGuru,

On the callcentric website, is there any way for me to see how much they billed me for each call, or for a given time period? I can see how much my balance has gone down, then subtract off the fixed incoming line service charge, and get the amount I spent, but it'd be nice to see the charge for each call, or for each day, etc.

On another note, forget porting a local number. It doesn't only cost $25. It also costs however long it takes for the port to complete, as you need to keep your old number active and in service until the port is complete, and all this info has to be exactly right, and you have to sign forms and scan them and upload them to your account, as well as a copy of your bill showing your full address (which gets torn off and is gone once you've mailed in your payment).

So, I was like, screw this. Too big a hassle just to keep the same number. I'll just call and cancel my old number and service outright.

Oh, and I think changing those port gains from -3 to -5 made a definite improvement. So far it has completely eliminated echo and the sound is improved. No clipping at all. Although, 95 percent of that went away with the firmware upgrade. I also think that DMTF (or whatever that is) is better set at Auto than AVT. As you suggested in the guide, of course.

[ElectronGuru]

Dude, you're like done! Though if you can have another problem, we can find even more useful **** codes

To view call history, log into your account and at the end of the second menu row, click Reports. Then set your date range (or click one of the preset ranges) and other filter options before clicking View Report. If you leave Call Direction on All, pay particular attention to the Destination column of the report. Monthly presents are also available, but hard to find but you can also do it manually (YEARMO):

https://my.callcentric.com/call_history/200904

And at the bottom are the happy totals:
589 min $4.3582 $0.0000 $4.3582

[js]

Thanks, ElectronGuru!

On another note, my brother got interested in doing this as well, and he did that VoIP test (not the speakeasy one, but the more comprehensive one) and these were his results:

QoS was 1 percent (!)
jitter was 1635.1 mS
packet loss was 16.9 percent.

Supposedly his download speed is 11000 or so kbps, as measured using speakeasy, but his download speed was 1.65 mbps using the VoIP measure, and his upload was 305 kbps.

I could NOT frigging believe how bad these numbers were. This can't be right. Either his router is hosed, or he has a bad cable, or his cable modem (because he has high speed cable internet) is hosed, or his ISP is way off their game.

Am I right in being simply flabbergasted by these numbers? I mean, a 17 percent packet loss? That's outrageous! And a 1.6 second jitter? WTF?

Right?

[js]

Oh, and one other question. If your connection is assymmetrical, like 2 Mbps download and 1 Mbps upload, which figure would you use for the QoS 80 percent? I presume you would use 80 percent of the download speed, as very little uploading is normally taken up by most users. However, on the other hand, the field into which you put that number is lableled "Maxiumum uplink speed".

What is the deal?

*edit*

I just reread the first post in this thread, and it does specifically mention "upload" total as the figure for calculating the 80 percent QoS number. I'll just go with that. (I just increased my connection from 1 mbps / 1 mbps to 2 / 1. For only an extra $10 per month. I figured that with all the money I am saving from not paying $65-$70 / month on the phone bill, I can well afford it.)

[cjames]

I've now got my (SPA-2102)set up in the same way as ElectronGuru does.


(adsl modem > timecapsule > spa-2102)


As I mentioned above timecapsule has nat/dhcp on; serves 10.0.x.x addresses to two laptops. I simply plugged a spare LAN port on the timecapsule into the WAN port on the SPA-2102. Turned it on and everything worked!

I'm happy that everything went so smoothly but still a bit curious. According to the phone *codes the SPA-2102 has an ip address of 10.0.1.3 (or 10.0.0.3? can't quite remember at the moment...) but I can't access the web-based setup page if I put this address into my browser.



ElectronGuru I'm still interested to know if you have any QoS settings enabled on your modem/timecapsule for the SAP-2102, if you can access the web-based SPA-2102 page the way you have things setup and if you have any other hints for using it with a timecapsule.



Any comments appreciated!

[ElectronGuru]

Not sure how I missed your first post, but

I'm curious how you found the thread (this guide), was it some kind of Google search?

The biggest criteria for deciding which goes first, your exiting router or the 2102, is whether you need QoS and if so, whether your existing router has the option. QoS is the one key feature Apple routers don't have, so the next question is, how fast is your connection? If its fast enough not to need QoS, then leave the Time Capsule first in line. Whatever the rate, there's no harm is just trying the 2102 second as you've done, and going on your mary way.

I was wondering what settings I should fill in to have it setup as you do above. As things are now I have:
I assume I plug an ethernet cable into the lan port of the timecapsule and in to the wan port of the spa-2102. I assume that I don't have to worry about the PPPoE settings on the spa-2102 and that I'll need to set wan to DHCP and make sure that the DHCP server on the spa-2102 is disabled.

Almost right. The trick here is that home networks typically have 1 router (used to have 0) and this essentially creates 2. If the TC is dispensing DHCP, then the 2102 is receiving its WAN side IP from the TC. In this configuration, the 2102's own DHCP server is only active on its LAN port, to which nothing is probably connected. So off or on, its not doing anything, so turning it off is only something you'd have to reverse if it ever changed it to first in line.

Are there any other settings i shoudl know about? Do I need to worry about setting QoS on my router or forwarding ports on the router and/or timcapsule?

In the simpler downstream configuration, the QoS isn't being used (and has no effect), and neither do any of the forwarding options need to be configured. Any forwarding in place on the TC should continue to function as before. The advantage here of the 2102 is not that you need QoS but that you may in the future and would already have it when needed.

As I mentioned above timecapsule has nat/dhcp on; serves 10.0.x.x addresses to two laptops. I simply plugged a spare LAN port on the timecapsule into the WAN port on the SPA-2102. Turned it on and everything worked!

Nice!

I'm happy that everything went so smoothly but still a bit curious. According to the phone *codes the SPA-2102 has an ip address of 10.0.1.3 (or 10.0.0.3? can't quite remember at the moment...) but I can't access the web-based setup page if I put this address into my browser.

This is the double router situation playing tricks again. The default config assumes the 2102 is directly on the internet. To protect itself, this config prevents access to the setup page via the WAN port (the otherwise correct way its currently hooked up). What you need to do is turn off this security option. Fire up this phone code:

7932# [plus password# if set], then press 1# (enable access via WAN port)

..enabling WAN side access (telling the box that both sides of the router are on a trusted network) and you should be able to get in.


Does that help?


More soon JS, tonight Lego's took priority:
http://www.candlepowerforums.com/vb/...d.php?t=235256

[ElectronGuru]

he did that VoIP test (not the speakeasy one, but the more comprehensive one) and these were his results:

QoS was 1 percent (!)
jitter was 1635.1 mS
packet loss was 16.9 percent.

Supposedly his download speed is 11000 or so kbps, as measured using speakeasy, but his download speed was 1.65 mbps using the VoIP measure, and his upload was 305 kbps.

Here are my numbers from the same tests:

speakeasy
30402k
5862k


voip review
Download speed 5005 Kbps (socket test)
Upload speed 4084 Kbps (socket test)
Quality of service 88 %
Maximum delay 112 ms
Round trip time 131 ms
Upstream jitter 0.9 ms
Upstream packet loss 0 %
Upstream packet order 100 %
Upstream discards 0 %

My SpeakEasy is also faster. A specilized test like this doesn't need to go past a few thousand K and the voip review guys probably didn't build it to do so. After getting the first page of results, have him click the DETAILED button to get the extra numbers and graphs. His numbers are bad and probably generated some non green dots on the page 1 report. Here's my jitter graph from page two:

I could NOT frigging believe how bad these numbers were. This can't be right. Either his router is hosed, or he has a bad cable, or his cable modem (because he has high speed cable internet) is hosed, or his ISP is way off their game.

Am I right in being simply flabbergasted by these numbers? I mean, a 17 percent packet loss? That's outrageous! And a 1.6 second jitter? WTF?

WFT indeed. As you suggest, there are a variety of variables here. Lets start with the two largest, the stuff he can't control (ISP, neighborhood, outside wiring) and the stuff he can:

1. How old is his cable box?
2. When was the last time all the network hardware was power cycled?
3. How long is the cable connecting the box to the wall?
4. How old is that cable?
5. How thick is that cable?
6. Is the cable modem the only device connected to that jack?
7. Are there any other devices connected to any other cable jacks in the house?
8. How many and which devices are in between the cable modem and the testing computer?
9. How old is the testing computer
10. How old is the java software on the testing computer?

[js]

EG,

I'll ask him these questions.

On another note, it seems to me that your guide should also say to set QoS Dsq to "TBF" = total bucket filter, right? Tom's Hardware reviewed the SPA2000, with an earlier firmware version, and he found significant improvement with TBF enabled.

Also, why should the second preferred codex be 729a, and not, say 726-40? I'm trying this out, with the third codex being 726-20. 711u is 64kbs encoding, high quality, and 726 is high quality as well but doesn't put as much bandwidth strain on things, depending on the kbps rate you chose. 729a is certainly easy on bandwidth, but it outright sucks at sound quality.

Also, what is your DTMF holdoff method? "strict"? And what is your strict holdoff time?

[cjames]

Thanks for your reply and help!

I'm curious how you found the thread (this guide), was it some kind of Google search?

Yes exactly this! I searched for SPA-2102 and timecapsule figuring someone out there had done it before.

What you need to do is turn off this security option. Fire up this phone code:

7932# [plus password# if set], then press 1# (enable access via WAN port)

..enabling WAN side access (telling the box that both sides of the router are on a trusted network) and you should be able to get in.

Thanks for the help. Unfortunately my isp seems to have set a password. I spent about 15min on hold tonight trying to get through to their helpline before giving up. None of the default codes I googled seemed to work.

I'll try again later I guess.

More soon JS, tonight Lego's took priority:
http://www.candlepowerforums.com/vb/...d.php?t=235256

Nice depth of field on that "dramatic" shot!

[js]

ElectronGuru,

I don't think that your recommendation for how to set the QoS maximum uplink kbps number is correct. According to this qos tutorial the number should be set at 90 to 95 percent of your lowest uplink speed. Now, this is the case if that figure, that setting, is used by QoS to properly fill the pipeline with packets. It needs to know how much bandwidth it has to fill in order to properly shape the upstream traffic. Some routers automatically and dynamically figure this out, but it seems that reports are that they don't do this well, and setting it manually to 95 percent of your lowest uplink speed is best.

On the other hand, if that figure in the 2102 is only limiting the input from the ethernet port to the internet port, and leaving all the rest of the bandwidth free for the VoIP to use, then 80 percent is the number to use--or rather you'd really want to figure out how much kbps you need for your particular codex, I would think, and subtract that amount, plus a margin, from your uplink speed.

But, my experience thus far makes me suspect that lowering the QoS uplink speed does not free up more uplink bandwidth for the 2102 to use. It didn't help at all, in fact.

Second, I also suspect that unless you chose QoS QDisc = "TBF" (instead of "NONE"), you simply won't have QoS working for you. It has to have some method to shape the upstream packets, and "NONE" is, as it says, no method at all = no shaping at all. But, I'm just guessing on this, and am still in the process of testing it out in real world use.

[js]

So, I just remembered something: when I lowered the uplink number, and tested during a phone call, the uplink speed my computer saw was approximately that uplink parameter minus a 200 or so kbps. So, I more than suspect that this paramter is TOTAL LAN to WAN throughput, including the 2102 VoIP throughput. So, you want it set to the lowest uplink speed you are likely to see, minus 5 percent or so as a safety margin.

Next, DTMF is "Dual Tone Multi Frequency" (or something like that), better known as "Touch Tone" (as opposed to Pulse--i.e. old rotary phones). So, it is all about getting the numbers you press on your keypad to do what you want them to do. AVT (also known as RFC2833) is an out-of-band method. IN-BAND, is, of course, an inband method. This would only work with codex 711u, as 729 doesn't have enough fidelity to transmit these tones well. Callcentric suggests setting the DTMF Tx method to "Auto", although I have had no issues with it set to AVT, either. I currently have mine on Auto. However, if you do have auto set, make sure to have a "yes" for AVT and in-band process settings. (I think those are the ones--those are the defaults, in any case). AVT is preferred overseas, if I'm remembering my recent reading correctly.

In any case, I'm not sure how these settings relate to voice sound quality. I'm sure a beep or two could get generated in the middle of a call, maybe, but other than that, I'm still researching this issue.

[js]

I found this and thought it was worth posting:

DTMF Tx Mode

DTMF Detection Tx Mode is available for SIP information and AVT . Options are: Strict or Normal. The default is Strict for which the following are true:
•A DTMF digit requires an extra hold time after detection.
•The DTMF level threshold is raised to -20 dBm.
•The minimum and maximum duration thresholds are:
• strict mode for AVT: 70 ms
• normal mode for AVT: 40 ms
• strict mode for SIP info: 90 ms
• normal mode for SIP info: 50 ms


The ATA Admin Guide also mentions another feature which I don't find on my SPA3102:

DTMF Tx Strict Hold Off Time:
This parameter is in effect only when "DTMF Tx Mode" is set to "strict," and when"DTMF Tx Method" is set to out-of-band; i.e. either AVT or SIP-INFO. If a user inadvertently sets the value to less than the default value, the system checks and reverts to the default value. There is no max limit on what the user can set of this parameter. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of dtmf detection, which is needed for interactive voice response systems (IVR).
Default is 90ms.

Main point is that if you set mode to strict, make sure the strict hold off time isn't below the minimum listed for the DTMF Tx Method you might use. If you have Auto set, chose the greater of AVT or SIP.

[ElectronGuru]

Outstanding JS, you are officially going past where I've been. I'll follow and update the OP as needed.

Also, why should the second preferred codex be 729a, and not, say 726-40? I'm trying this out, with the third codex being 726-20. 711u is 64kbs encoding, high quality, and 726 is high quality as well but doesn't put as much bandwidth strain on things, depending on the kbps rate you chose. 729a is certainly easy on bandwidth, but it outright sucks at sound quality.

Everything I’ve read focused exclusively on codecs ending in letters and most describe 729a as the best sounding but the most bandwidth intensive. My tests showed this as well. Let us know what you find out. Keep in mind, if Call Centric doesn't support a codec, it will be a short test.

On another note, it seems to me that your guide should also say to set QoS Dsq to "TBF" = total bucket filter, right? Tom's Hardware reviewed the SPA2000, with an earlier firmware version, and he found significant improvement with TBF enabled.

Second, I also suspect that unless you chose QoS QDisc = "TBF" (instead of "NONE"), you simply won't have QoS working for you. It has to have some method to shape the upstream packets, and "NONE" is, as it says, no method at all = no shaping at all. But, I'm just guessing on this, and am still in the process of testing it out in real world use.

I’ve not had ‘little enough bandwidth’ to know for sure. I think you’re right, and have updated the OP.

I don't think that your recommendation for how to set the QoS maximum uplink kbps number is correct. According to this qos tutorial the number should be set at 90 to 95 percent of your lowest uplink speed. Now, this is the case if that figure, that setting, is used by QoS to properly fill the pipeline with packets. It needs to know how much bandwidth it has to fill in order to properly shape the upstream traffic. Some routers automatically and dynamically figure this out, but it seems that reports are that they don't do this well, and setting it manually to 95 percent of your lowest uplink speed is best.

On the other hand, if that figure in the 2102 is only limiting the input from the ethernet port to the internet port, and leaving all the rest of the bandwidth free for the VoIP to use, then 80 percent is the number to use--or rather you'd really want to figure out how much kbps you need for your particular codex, I would think, and subtract that amount, plus a margin, from your uplink speed.

But, my experience thus far makes me suspect that lowering the QoS uplink speed does not free up more uplink bandwidth for the 2102 to use. It didn't help at all, in fact.

when I lowered the uplink number, and tested during a phone call, the uplink speed my computer saw was approximately that uplink parameter minus a 200 or so kbps. So, I more than suspect that this paramter is TOTAL LAN to WAN throughput, including the 2102 VoIP throughput. So, you want it set to the lowest uplink speed you are likely to see, minus 5 percent or so as a safety margin.

There is contradictory information on the "true meaning" of Maximum Uplink Speed. My best understanding is that is tells the 2102 the total possible upload of both voice and data that is available. I don't see why they wouldn't design it to exceed this number, so think of it as Expected Total Upload Speed.

Also, what is your DTMF holdoff method? "strict"? And what is your strict holdoff time?

Next, DTMF is "Dual Tone Multi Frequency" (or something like that), better known as "Touch Tone" (as opposed to Pulse--i.e. old rotary phones). So, it is all about getting the numbers you press on your keypad to do what you want them to do. AVT (also known as RFC2833) is an out-of-band method. IN-BAND, is, of course, an inband method. This would only work with codex 711u, as 729 doesn't have enough fidelity to transmit these tones well. Callcentric suggests setting the DTMF Tx method to "Auto", although I have had no issues with it set to AVT, either. I currently have mine on Auto. However, if you do have auto set, make sure to have a "yes" for AVT and in-band process settings. (I think those are the ones--those are the defaults, in any case). AVT is preferred overseas, if I'm remembering my recent reading correctly.

In any case, I'm not sure how these settings relate to voice sound quality. I'm sure a beep or two could get generated in the middle of a call, maybe, but other than that, I'm still researching this issue.I found this and thought it was worth posting:

Main point is that if you set mode to strict, make sure the strict hold off time isn't below the minimum listed for the DTMF Tx Method you might use. If you have Auto set, chose the greater of AVT or SIP.

I messed with DTMF Tx Method trying to get my faxing to work when suddenly automated phone menu systems couldn't hear by number beeps. Along a similar line, there are settings in the ring section that incorrect can block an answering machines ability to count rings. I agree, this has nothing to do with sound quality.