Chattering on Fxs2


#1

Greetings.

I finished setting up a two line…one DID…system using Grandstream HT802.
FXS1 is a phone coming straight out of the ATA. FXS2 is a line to a jack and then a dry loop to the kitchen using black and yellow. There aren’t any lines coming off this wire which has a reasonable guage. The length of this span is about fifteen feet. I configured FXS1 and 2 identically using the same numbers throughout.

For about a day whenever the phone on the dry loop was picked up there was a chattering effect on the dial tone. It sounded slightly intermittent. A call would be slightly choppy as well. Today it stopped doing this and is completely normal. I’m planning on going on a trip and don’t want to leave my lady friend with a line that may revert to this dysfunctional chattering so I have a few questions.

I noticed on the setup of FXS that the default from one to the other on a few entries is different. Could this be causing the discordance. Can FXS1 and 2 be identical.

For example default Local sip port on FXS1 is 5060 on 2 it is 5062 (voip.ms prefers 5080)
RTP port is 5004 and 5012 respectively. (voip.ms prefers 10000)

or

could it be that GAIN is an issue
I have TX -2dB and RX -2dB

Perplexed.

These tests may indicate something…still a newbie at all this:


#2

IMO, you would be better off using a single FXS port and using a splitter/duplex adapter on the line to feed both devices.

The chop is most likely an internet issue called jitter. This is where packets are not arriving at the HT at the same rate as when sent and perhaps the packets are not in the same order.

Gain is used to adjust the volume heard at the remote end (TX) and at your end (RX) and has nothing to do with the issue.

I do not know that the provider will support two different registrations to the same account from the same IP, where each is using a different port.


#3

Thank you for the reply lpneblet.
Apparently voip.ms will support multiple fxs entries per DID. I suppose I’ll have to contact them about configuration values between each if there are differences.

The jitter you mention would be extremely annoying if repeated making a phone call unenjoyable. The thought of having to try out different internet providers until one sends a steady stream of packets seams overwhelming. I talked with one of the technicians about the minimum bit rate needed for voip and it was mentioned that voip has been around for a while even when 256 kb/s was high speed for some and it worked well then.
I don’t know how to respond to this jitter phenomenon in this day and age. Shouldn’t internet providers have it together by now.

If there is a return of the jitters though I’ll consider the splitter you mention. The point of getting a two port ATA was to prevent having devices like this on the line. Some adapters I’ve seen are poorly made with stranded wire inside rather than a solid wire with a decent gauge.


#4

Jitter is a form of latency and can be introduced anywhere in the path. It may not be the ISP but could be on your end or from one of the carriers in between you and VOIP.ms. If you have other devices in the home that make use of the Internet, then consider for a moment that you have the connection with the ISP. This connection is likely the slowest link between you and the intended destination as I assume the internal home network is faster than what your ISP is providing. In the meantime, all the devices on your network are considered peer unless you have some form of QoS or other classification scheme. Depending on the codec in use, the voice stream will use less than 100kbs, which in the scheme of things is quite small. If using g729, much less than 100kbs. However, unless there is some form of prioritization or bandwidth reservation, then all packets from all devices are treated the same. The voice stream can be impacted by the other devices that need more bandwidth which results in the router trying to push X amount of data into the ISP when the subscription only supports Y. As a result, the router may not be able to deliver the data smoothly. You may need to examine your usage and see if QoS would help so that the voice packets are prioritized.

The Internet is a dynamic thing which when using voice it is real time which is not the same as HTTP, or most other requests to include streaming where most of these have some form of error correction or buffering. In voice you have an equal amount of data flowing in both directions and any packet that is dropped or arrives out of sequence or with timing variations between packets may cause the users to experience some issues.

Perhaps the above may not be the case in your situation, but I am only pointing out the possibilities for consideration of any action that you can take to better bullet-proof the function. I have VoIP at the house using an ATA and have had little issue over the years, Of course, the ISP has had some plant issues, but these have been fixed in relatively short order and I have it set such that if the provider does not see a registration come in from the ATA, they will wait for one more expiry period and if still not seen will automatically invoke directing the calls to my cell phone, When they see a register, they will then re-direct back to the ATA. It takes a total of 4 minutes as the expiry is 2 minutes.

The splitter/duplex has nothing to do with the jitter issue, but rather the ease of distribution of the analog phone signal to a couple of devices without the need to have two registrations for a single DID. If the provider does indeed allow multiple registrations for a single DID, then the only difference between FXS1 and FX2 would be the local SIP port. These need to be different to help the router maintain a proper NAT/PAT so that the messaging from the provider does not get mixed up, To the provider, you are contacting them as single IP (your public), but the only things that lets them know what message goes to which port on your device is the local SIP port. As the two ports are indeed different in the respect that you have one phone connected to one port and the other phone to the other port, they need to know to which port/phone to respond. Hopefully, you have the router set to forward correctly and that the SIP ALG is off in both the router and the ISP modem.


#5

It has been quite busy for me getting ready for my trip which I’m on now and setting up the ATA before I left. I found out at the last minute that the downstairs line wasn’t actually ringing being on fxs2. I made a subaccount and ring group and there were still issues so I went with your suggestion to put it all on fxs1 which worked. I’m not there any more to monitor how it is going. I was hoping to eliminate jack splitting adapters for the reason mentioned but it is working.

There is still the jittering on the house line which lasts about ten seconds and goes to normal. I guess we’ll just have to live with it. As this is the original reason for creating this post I can’t say this issue was resolved but I appreciate the replies.

Regarding FXS 1 and 2…it would be nice to see an adjustable setting in the grandstream configurator to tie these together as one so it can function like a normal household with ease. I personally don’t mind spending days on configuring things which is why I enjoy Linux but when you have a third person involved who wants it done yesterday then easy outcomes are preferred.


#6

If it always goes away and stays good for the duration of the call, then it seems unlikely an internet issue, but something else. If a network capture is possible, then maybe more can be found out. You might also try setting the codecs to g729 and then g711 as the first settings in the preferred vocoder list. Then enable use first matching vocoder. See if this helps with the first few seconds and you can try and change by setting g711 and then g729 (reverse the entries) and see.


#7

I went with pcmu then g729 followed by g711. I was a bit hesitant with pcmu as I read that this codec requires the most bandwith and I didn’t know if the line would become choppy from competing usage. One thing that I’ve noticed during speed tests is that during the upload test it takes five to eight seconds to activate the server and then springs into action. This seems to be the same amount of time for the chattering effect of the dial tone before going to normal. May be a coincidence as it seems the Grandstream would be sending the dial tone.

I wont be back for a while to test out your suggestion but I’ll be sure to reply again with an update if the vocoder eliminates this issue. Thank you for all the replies.


#8

Ohhhh, wait,.,

I missed that part (dial tone) completely as I was focused on the voice. I am so sorry.

If, when you first pick up the receiver and are greeted with a dial tone that is interrupted for the first few seconds and then suddenly stops and the dial tone is fine, that is called a stutter tone and is indicative of you having voice mail. This is done so that those with phone without a display and/or for those with vision impairments who are unable to see a display or visual indicator, will know there is VM pending.

You should check with the provider about it.

https://wiki.voip.ms/article/Voicemail