They decided instead to send 30 channels down a line instead of 23 and call it E1, presumably because the lines were in better shape.
E1 is a European standard and runs at 2Mbit instead of 1.544Mbit.
But having said that, maxtherabbit's experiments (over the internet) do confirm you can do a modem call over SIP. So perhaps it's the good-enough area again.
Oh, there are entire ISPs these days that run on VoIP if you want.
I remember that when the unannounced changeover happened I couldn't dial with the pulse rating set at 38 (IIRC default/standard = 70) with my 14k4 modem, and a bit of experimentation showed that 40 was the lowest value that worked. Or perhaps this was how fast DTMF tones could be sent?
That only affects pulse dialing. If your make/break ratio is wrong, electromechanical ORs won't step correctly.
Most, probably all, electromechanical exchanges got add-on tone-to-pulse converters and I think it was faster to use DTMF and let the converters do their thing than pulse dial myself, or maybe I just imagined that.
That sounds about right. Tone dialing is much faster, and SXS ORs could be stepped really fast with the right setup (usually used for trunks).
I wonder if it's possible anywhere to connect via VoIP and achieve better sound quality than what 64kbps 8kHz uncompressed can do? I.E. is it possible to use any of the better sounding cell phone encodings over VoIP?
I'm not sure about VoIP specifically but you can most definitely do high bitrate audio between certain brands of cellphones.
I think it's a bit confusing that there is ATM, T1/E1 and ISDN PRI which seems to be somewhat the same thing, but perhaps not really fully, kind of sort of, perhaps.
ATM is a little bit different. T1 and E1 are essentially US and Euro versions of the same standard. PRI is a protocol that runs on top of a T1 or E1 link.
IP. The way I've been testing is to register each FXS port to a different subaccount on voip.ms and then use the internal extension number to call one from the other. So it's still a "loopback call" in the sense that I'm calling my own equipment but it is traversing the voip.ms server.
As a baseline, I did the same test from my premise through the same voip.ms server using MicroSIP (softphone app which logs latency and jitter metrics) and my desktop Cisco IP phone. Round trip latency was 10ms. I don't know if MicroSIP includes the decode latency in this metric (I suspect not) but this at least tells us that the packets are not spending an unreasonable amount of time in da cloud.
I would think this would be QoS, but if it really is only 10ms for the packet delivery, your ATA is really really slow. Not sure why that would be.
I made a lowball offer on a Patton SmartNode 4112 on ebay last night and the guy accepted it. So I'll be able to compare that one soon. I did look for any cheap TA900 series stuff, but there was nothing available right now that wasn't broken.
Fair enough, best of luck with it. Hope it doesn't need manuals or software, because Patton is ridiculously tight-fisted with that stuff.
I am able to connect to WebTV using the original box with its dial up modem on Comcast land line.
Nice. I have not been able to get into ReDialed because their primary number is intermittent at best and there are exactly zero good instructions for making it work on a local RAS, or in Windows with PRI cards. The Discord is mostly full of children with RPis and USB modems with a battery, and the few people who seem to actually know what they're talking about have never seen a Windows install in their life, apparently. Someday I'll get mine working.
The Patton Smartnode 4112 is nice. I'd say for anyone wanting to setup a basic analog ATA for modem use this is definitely the one to get. Much better choice than the Cisco ATA191 or Grandstream HT802.
That being said, it is a professional device and that shows. Both in the quality and in the learning curve for setup. If anyone is interested I can post my config file for use with voip.ms.
Please do post your config. Did it have a web UI setup or was it all command line?
I think I'm going to look into Anveo direct though, they don't proxy the media stream like voip.ms does - it's a direct RTP connection to whomever terminates the calls, which should reduce latency and jitter even further.
Let us know how that turns out, I'm still looking for a VoIP provider. Have you looked into The Serial Port's VoIP phone system? Being set up by hobbyists specifically for modems might help your chances.
Next step is to start looking for a deal on an upgraded Patton which has FXS and BRI. They seem significantly less available on the secondary market than the FXS-only models
Wait, there are BRI ones? I would really like to get some BRI ports on my phone network. I've got PRIs and FXS but no BRI since that is unusually difficult to get economically. I've got a Teltone ILS-1000 so I can actually make connections, but it's rather inflexible and I have no way to tie it into the rest of my network.
How about a DIY solution using a sound card and some analog circuitry (plus a few I/O pins) to make it act as a phone line?
You could emulate two lines with every sound card as it's in stereo, and for those "home studio" type cards you could emulate more lines. The latency would just be whatever the PCI bus and whatnot introduces, i.e. negligible. I don't know how hard/easy it would be to write software for it to interface with for example Asterisk. The analogue circuit is fairly easy. Either use a transformer or OP amps for echo cancellation, or just ignore that and do it fully in software, and then just a voltage/current detector to indicate on/off hook to the computer, and an output that drives a relay that connects ring AC voltage, plus a circuit that immediately disconnects the ring voltage if the connected phone/modem goes off hook, and also a circuit that disables the ring signal as long as the connected device is off hook. Perhaps add some sanity check for the ring signal to avoid it being switched on by the wrong software (if using some generic port for the control signals).
Feel free to attempt it. I've never heard of this being done, personally, but I don't really see a good reason why it couldn't be done? Just sounds like a lot of work to me at least. If you get one up and running that will work under Windows, I'd be happy to give it a try and help debug.
Especially some of the home studio type cards can sync their clocks via S/PDIF and thus you can sync up a bunch of cards to run at an identical clock, and if you also use some E1/T1 card or whatnot with Asterisk you could also take the clock from that card somehow and use it to generate an idle S/PDIF signal if you want everything to use the same clock.
A synced clock is absolutely essential for any of the digital telephony connections. Sadly, Asterisk does an absolutely deplorable job at handling TDM, compatible cards are hideously expensive, and you would still need to generate your clock somewhere else as most of those cards can only act as a TE, not as NT. I've seen a few that can act as NT, but they usually require an external clock source. I've seen people using GPS clock sources for T1s, though. Maybe that's worth a look.
