More analog modem advice. HT802


#1

I have an embedded septic system telemetry controller with an analog modem adapter. The service company wants $700 to switch to an ethernet adapter, and I would need to run additional wiring. A POTS line in my area is $30 a month.

Then I’d like to give VOIP a shot. Reading the manual the device has a serial port which you can connect to at 9600 baud, and then I think the existing modem adapter might be running at this speed.

There seem to be posts here which imply that some people got data calls working with HT 702: Analog Modems on HT702

Furthermore there are some other suggestions:

Then I’m a VOIP novice but interesting in what configurations I should try to see if I can make it work.

Finally I will have to choose a SIP provider. Then I assume I’d be best off finding one nearby, but maybe there is some criteria I should be looking for when selecting a provider for test.

Any suggestions or advice would be appreciated.


#2

There are no guarantees.

A modem use various protocols that form standards by which communications can be had between two similar devices over a standard POTS line. As the line conditions change, the modems can oftentimes re-adjust to accommodate.

An ATA attempts to emulate a POTS/PSTN line, but the analog stream from the modem is converted into data packets which are encoded using a voice codec. As your profile shows to be NA, you should use a G711u codec as this offers a wider frequency bandwidth. As the modem is using audio frequency tones (much like a fax) to communicate, the need is to use a codec that will support frequency bandwidth requirements such that tones are not compromised as they go from point A to B.

Now comes the tricky part.

The issues that arise are usually associated to the Internet carriage or local congestion and such. When traversing the network there are various issues that might arise that impair the delivery of the packets such that what was sent is not quite the same as what was received. Data packets from the sending modem may have looked like - 1, 2, 3, 4, 5, 6, 7, etc., but at the receiving end may look like - 1…2, 4,3, 5…6, 7. They may not arrive at the same timing interval nor may they always arrive in the same order and in some cases, some packets may not arrive at all. Most of the time, this is not an issue as carriers and ISPs have gone to great lengths to improve the networks as they recognize the growing significance of VoIP and similar technologies. The use of jitter buffers in the ATA may help on the receive end of things, but you still have to insure that when sending there is enough bandwidth available for all devices on the network such that the upstream is sent without issue. You may very well need to employ QoS on your router and ATA (I would anyway).

If it were me, I would base the decision on moving to VoIP on how mission critical is the device being monitored and the impact that it may cause if it does not work when most needed. The Internet may go down, you have more equipment in the path that could fail, etc… This may be a different level of concern than trying to convert a credit card machine or postage meter, etc. If your current ISP has issues maintaining your service, then perhaps you need to reconsider. You also need to consider if you have a static or dynamic public IP and if public, then you should consider static and if not possible, then get DDNS with a FQDN.

If you elect to try, then find a SIP decent provider (read the reviews) and get a new number with the service (if the existing number is not pertinent to the monitoring). Keep the existing POTS line around until such time as you are satisfied that it all works.

Good luck.


#3

Hi thanks for the advice. The pots line was actually cancelled years ago and now we’re trying to get the system back online. It actually uses a programmable serial to ethernet adapter if we bought the hardware from the manufacturer. I have found said adapters online for about $30, but I assume they have custom firmware programmed. Then this isn’t public and so I’d have to spend $700 to get one programmed. Without programming I don’t know how it would know where to send the packets to. Also I believe there’s a password on the device when you try to access it via the serial port. Requires a bit of reverse engineering.

Then looking at the other side it already has the modem and is already configured to dial out to some modem bank. I’m in Western Washington and thought I saw a long list of VOIP providers in my area, so I will shop around and look at reviews. I wasn’t sure if a particular hardware setup on their end would make a difference.

I found goughlui.com/2016/11/20/teardown-review-grandstream-ht802-voip-analog-telephone-adapter/ which looks really well informed regarding this device. It actually swayed my opinion to go ahead and pick one up. He is in Australia so using another codec, but still really in depth.

Finally I have seen some claims regarding V.150 and V.150.1 devices having an advantage here. Some people recommend these devices: patton.com/solution/Fax-Modem-over-IP/ . It’s a little convoluted because there seem to be multiple ways to handle modem with VOIP. Some appear they might send the modem call is data and then use a modem to make the call on the other side? Others seem to imply better support for analog modem calls over IP. However these devices are at least 5X the cost of the HT-802, and what looks like an older hardware platform.


#4

I’m at the stage of testing this. I’ve signed up with flowroute, which seems to be a VOIP provider based in Seattle. I’m in the Western Washington region, so this seems to make sense. I have to go through a 3rd party that isn’t terribly helpful or responsive regarding test successes.

I am impressed with all the configuration options. Then are there particular configurations that might make a specific difference to the success of data modem calls?

I think some of the relevant ones might be found at the bottom of the page:

Voice Frames per TX: 2
G723 Rate: 6.3kbps encoding rate 5.3kbps encoding rate
iLBC Frame Size: 20ms 30ms
Disable OPUS Stereo in SDP: No Yes (removes “/2” from offer)
iLBC Payload Type: (between 96 and 127, default is 97)
OPUS Payload Type: (between 96 and 127, default is 123)
VAD: No Yes
Symmetric RTP: No Yes
Fax Mode: T.38 Pass-Through
Re-INVITE After Fax Tone Detected: Enabled Disabled
Jitter Buffer Type: Fixed Adaptive
Jitter Buffer Length: Low Medium High
SRTP Mode: Disabled Enabled but not forced Enabled and forced
Crypto Life Time: Disabled Enabled
SLIC Setting: USA 1 (BELLCORE 600 ohms) USA 2 (BELLCORE 600 ohms + 2.16uF) AUSTRALIA CHINA CO CHINA PBX EUROPEAN CTR21 GERMANY INDIA/NEW ZEALAND JAPAN CO JAPAN PBX STANDARD 900 ohms UK
Caller ID Scheme: Bellcore/Telcordia ETSI-FSK during ringing ETSI-FSK prior to ringing with DTAS ETSI-FSK prior to ringing with LR+DTAS ETSI-FSK prior to ringing with RP ETSI-DTMF during ringing ETSI-DTMF prior to ringing with DTAS ETSI-DTMF prior to ringing with LR+DTAS ETSI-DTMF prior to ringing with RP SIN 227 - BT NTT Japan DTMF Denmark prior to ringing no DTAS no LR DTMF Denmark prior to ringing with LR DTMF Sweden/Finalnd prior to ringing with LR DTMF Brazil DTMF-FSK Brazil
DTMF Caller ID: Start Tone Default A B C D # Stop Tone Default A B C D #
Polarity Reversal: No Yes (reverse polarity upon call establishment and termination)
Loop Current Disconnect: No Yes (loop current disconnect upon call termination)
Play busy/reorder tone before Loop Current Disconnect: No Yes (play busy/reorder tone before loop current disconnect upon call fail)
Loop Current Disconnect Duration: (100 - 10000 milliseconds. Default 200 milliseconds)
Enable Pulse Dialing: No Yes
Enable Hook Flash: No Yes
Hook Flash Timing: In 40-2000 milliseconds range, minimum: maximum:
On Hook Timing: (In 40-2000 milliseconds range, default is 400)
Gain: TX +6dB +4dB +2dB 0dB default -2dB -4dB -6dB -8dB -10dB -12dB RX +6dB +4dB +2dB 0dB -2dB -4dB -6dB default -8dB -10dB -12dB
Disable Line Echo Canceller (LEC): No Yes
Disable Network Echo Suppressor: No Yes

#5

I have a way to trigger a call manually on the telemetry device. Then I’m setting the trigger and use a simple wrapper around telnet to see if the line goes off the hook from the device to make the call.


#6

I assume you wish to know what options to set?

For the most part the default settings will work. You will want the first codec to be g711u/PCMU (they are the same thing, just depends on how listed). North American providers typically only support G711u and g729. You really want to use g711u as this is the wider bandwidth codec and has a better chance of passing the needed frequency tones. If possible, you may want to set all available codec to g711u so as to force the issue.

Symmetric RTP = No
Fax Mode = no or disabled, but really should not matter.
SRTP mode = disabled.
Jitter Buffer = is dependent on the line condition with regard to latency. I would start with fixed and 100ms and start with low and if needed move up to medium and high.
Both Echo canceler = Off
SLIC = USA1
CallerID=Bellcore
DTMF = leave what ever is set as the default
Polarity reversal = No
Loop current = Yes
Play busy= leave at default
Loop current disconnect - leave at current, but if the device will not dial out, you may have to increase setting.
Hook flash = no
Gain - leave at default

There are a lot of variables here whose settings are dependent on the connection. The settings I suggested with the exception of the codecs are pretty generic. You may want to hook up an analog phone and make a couple of test calls to see if echo and volume are OK. Keep in mind that the HT is using an electronic signaling mechanism rather than a mechanical mechanism as the phone. When you take a phone off hook, the mechanics are such that it will connect to the line pretty much regardless of the condition. The HT senses the line first and then if it detects, then it goes off hook and seizes the line, so you may need to play with the disconnect settings.
Stage dialing = 1
Dialing delay = 500ms.


#7

After running tests, I ran through the wiring and re-crimped some connectors.I see outbound calls going from the device through the provider dashboard. I now see some calls have lasted a few minutes. I will have to contact the 3rd party to see if they successfully posted the telemetry or not.

Thanks for the suggestions. Running through the configuration suggestions you have made:

I have previously set all the vocoder codecs to PCMU, which I believe corresponds to g711u.

I have now set both disable echo canceller and suppresor to Yes

I have set the Hook flash to No.

I have not yet set the jitter buffer settings. The defaults are set to:

Jitter Buffer Type: Adaptive
Jitter Buffer Length: Medium

I assume that fixed / low refers to the 100ms you are mentioning above. I should probably change the Jitter Buffer as you suggest. Just trying to not make too many changes at one time.

I’m also curious will the layer 3 QoS settings have any impact? Currently they are set as default for SIP and RTP…


Grandstream HT801 and Alarm
#8

QoS, most likely no impact. is this a dedicated internet connection to the HT or shared?