Calls Go Silent

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Calls Go Silent

Postby akasophistikat » 06/07/2021

Hello Community,

I had a Freephone ATA device HT-801 that I had setup. The phone worked as expected but then we started to experiencing calls going silent from 10 to 30+ seconds. Using these forms, I found and tried as many settings as I could to fix this issue but with no luck. Knowing that Freephoneline support was hard to come by, we decided it was best to switch to Fongo because the HT-801 came configured and the account came with support.

Unfortunately, the same issue exists. My ticket with support is still open as their first response was Bush League so I'm here to see if anyone has any advice.

Here's what I've tried so far.

* Different phones
* Different WiFi locations for our Satellites
* Freephoneline Account and a Fongo Account
* Two different ATA devices
* I looked online and improved my routers' settings for better connectivity and improved performance (based on articles from other Orbi user experiences')

Here's what's left for me to try.

* We have a second ATA device, using Freephoneline; temporarily disable this device and use call forward to my cell.
* Try a new location for our Satellites
* Give up and go back to Bell/Rogers etc., this issue is too consistent the people using this ATA device have lost the patience.

Thanks for any advice.
akasophistikat
Just Passing Thru
 
Posts: 2
Joined: 06/07/2021
SIP Device Name: Grandstream HT-801
Router: Orbi RBR20

Re: Calls Go Silent

Postby Liptonbrisk » 06/07/2021

akasophistikat wrote: their first response was Bush League .


I find I'm reluctant to assist since I wonder whether any attempt by me to help will result in potentially unwarranted criticism, and I'm not paid to assist.


But here is how I would look at the problem you're describing: the RTP audio stream stops reaching the ATA after 10 to 30 seconds.
Why?

Common causes might include

a) Firewall issue.
UDP Port used for RTP starts being blocked by NAT firewall.

You might want to check for a possible router firmware update.
https://www.netgear.com/support/product/orbi.aspx

You could port forward the UDP (RTP) ports used for the audio stream to your ATA, but it's a security risk.
Only port forward when all else fails. For assistance with port forwarding contact Netgear support staff, Netgear forums, or consult a manual.


b) ISP issue
Packets are intermittently being dropped along the path between the server and your ATA, or there's huge packet loss/jitter occurring. I would expect the call to eventually disconnect and not just become silent, however.

c) Router issue.

Attach the ATA directly to your modem using an ethernet cable, and bypass your router. Test.

d) Devices on your LAN are sucking up available bandwidth. QoS problem


1. What brand and model modem are you using?
If you are also using your own router in addition to the one supplied by your ISP, then you should be enabling bridge mode in it, provided it's a modem/router combo, hub, or gateway device. Contact your ISP for help.

2. Disable SIP ALG in your router. Here is an example: https://www.obitalk.com/info/faq/sip-alg/disable-alg. Netgear routers do have SIP ALG that can be disabled.

3. To help rule out SIP ALG problems, use voip4.freephoneline.ca:6060 in your ATA. For Fongo Home Phone, you need to make the request to be placed on an alternate server (one that does not use UDP 5060).

4. I would try to rule out your router being the issue and connect an ATA directly to your modem with an ethernet cable. If the problem disappears, you know where the issue lies.

5. A.Use winmtr https://sourceforge.net/projects/winmtr/ if you have a Windows PC.

B. For Freephoneline.ca (based in Ontario), 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.

C. Look at the very last hop or line. Take a look at your average ping--and your maximum. You want those values to be relatively close.
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 (pings over 200ms), you should probably avoid FPL.

I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca)-24ms (voip4.freephonline.ca) on average, depending on the server I'm testing to. Preferably, you want pings below 100ms.

Anything over 200ms is unacceptable.

What you don't want to see is 40, 45, 50, 35, 500, 40, 30, 45, 700. That's bad jitter, which leads to choppy calls.
You want relatively consistent pings without a lot of variation. Severe packet loss and ping spikes can produce silence and dropped calls.

I recommend reading page 42 under the choppy audio section from download/file.php?id=2162.
I also suggest reading the first 5 pages of the preamble.

6) Enable QoS for your ATA. Refer to router manual, Netgear support staff, and Netgear forums.

7) Again, you could try port forwarding the RTP (UDP) ports from your router to your ATA, but it's a security risk.
I would change local RTP port in the ATA to something other than default, and then I would have "Use Random RTP Port" set to no in the ATA.
It's easier and safer to port forward a single UDP port than a gigantic range.
However, I wouldn't port forward unless I had no other choice. This is something I would consider doing reluctantly.


I rapidly cobbled this post together while exhausted. I don't have any other thoughts at this time, but if you're with a major ISP, such as Rogers, you can ask them to check for signal strength, to check for noise issues, and to run periodic line quality tests to see if they notice any problems to your modem.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
Technical Support
 
Posts: 1840
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:8680EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: Calls Go Silent

Postby Liptonbrisk » 06/07/2021

If you're interested. . .

(Generic info)

Typically, for VoIP SIP services, especially for Freephoneline/Fongo, you want

A) a router that does not have a full cone NAT,

Visit https://dh2i.com/kbs/kbs-2961448-unders ... -punching/.
Mango from the Obitalk.com forums writes,
“Use a restricted cone NAT router, and do not use port forwarding or DMZ. Restricted cone NAT will only permit
inbound traffic from the service provider you're registered to. If you have a full cone NAT router, it will allow traffic
from any source. This is probably not what you intend.
If you have a Windows computer, you can test your router using the utility here:
http://www.dslreports.com/forum/remark,22292023. To run it, use stun stun.ekiga.net from a command prompt.”
Essentially, you download the stun-test.zip file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit
http://www.thewindowsclub.com/how-to-ru ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit
http://www.digitalcitizen.life/command- ... c-commands); and type “stun stun.ekiga.net” without
the quotation marks followed by the enter/return button on your keyboard.
Asus routers, at the time of this writing, produce port restricted cone NAT routers, for example and are fine,
provided you’re using one with Asuswrt-Merlin, third party firmware installed.

B) a router that lets you disable SIP ALG if it's buggy,

To understand why SIP ALG often causes horrible problems, please visit
https://www.voip-info.org/routers-sip-alg/ (scroll down to the section on SIP ALG problems).

If you're dealing with a modem/router combo issued by an ISP or a router with SIP ALG forced on, you may have
to use voip4.freephoneline.ca:6060 for the Proxy Server. The purpose of voip4.freephoneline.ca:6060 is to circumvent
faulty SIP ALG features in routers.

C) a router that allows you to set QoS or assign highest priority to your ATA or IP Phone over all other devices on your LAN (local area network),

For a very general description of what QoS can do for you, visit https://www.voipmechanic.com/qos-for-voip.htm.
The basic idea is if you're torrenting or have a bunch of other computers, smartphones, tablets, etc. downloading and uploading (hogging all your available bandwidth), you don't want
your ATA not to have access to enough bandwidth to make or receive calls properly. So QoS or a Bandwidth Monitor feature (which is just another form of QoS) is a really good idea for VoIP users.

I often get an occasional relative complaining to me, "Hey my calls sound choppy." And then when I go visit, some kids are playing MMOs on a computer, while another person is downloading a huge file,
and another person is backing up files to a cloud service all at the same time someone else is trying to talk on the phone. All those devices, without QoS enabled, are fighting over available bandwidth along with the ATA.

and D) A router that lets you adjust both Unreplied and Assured UDP timeouts.

Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working properly with a VoIP SIP provider like Freephoneline, in particular after power failures, the following conditions must be met:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires; for Grandstream, the setting is SIP OPTIONS Keep Alive Interval) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (or RegisterRetryInterval in Obihai ATAs)

“<“ means less than.

When a modem leases a new IP address, a problem can arise where prior associations using the old IP address are maintained in the router. When the ATA attempts to communicate using the old IP address, the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval (and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where the corrupted connection persists. If UDP Unreplied timeout is, for example, 15, and the NAT Keep Alive Interval is 20, then the corrupted connection will timeout or close. A new connection will be created, and everything will work fine.

Another problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often 180 by default in consumer routers): the NAT hole will close due to the ATA not communicating frequently enough with the SIP server. In turn, incoming calls may, intermittently, not reach the ATA. Again, X_Keepalives expires is supposed to be 20 with FPL.

(the above settings are making reference to those in Obihai ATAs)

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin (I would avoid any model below/less powerful than an RT-AC68U), third party firmware for Asus routers, does offer easy access to these two settings, which are found under General–>Tools-->Other settings. My understanding is that third party Tomato firmware has these two settings as well. So if your router supports Tomato firmware, that may be another option. Note that I will not be held accountable any damage resulting from failed firmware updates. Apparently, Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well: https://forums.redflagdeals.com/recomme ... #p28059363.

Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT


The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 15 for UDP Unreplied Timeout and 115 for UDP Assured Timeout.



ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
User avatar
Liptonbrisk
Technical Support
 
Posts: 1840
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:8680EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: Calls Go Silent

Postby akasophistikat » 06/08/2021

I find I'm reluctant to assist since I wonder whether any attempt by me to help will result in potentially unwarranted criticism, and I'm not paid to assist.

I received a copy and paste version of "Have you restarted your modem?" and for a paid service, I was expecting what you provided. In fact, you provided more than what I was hoping for so I'll start by saying thank you for your response.

I already took steps to improve my connectivity and will go through your notes to see what else I can do. I've been visiting Netgear's forum as well and will continue to keep an eye to see if anyone has had this problem. If I can find a solution I'll be sure to post all the steps I took.
akasophistikat
Just Passing Thru
 
Posts: 2
Joined: 06/07/2021
SIP Device Name: Grandstream HT-801
Router: Orbi RBR20

Re: Calls Go Silent

Postby Liptonbrisk » 06/08/2021

akasophistikat wrote:I already took steps to improve my connectivity and will go through your notes to see what else I can do. I've been visiting Netgear's forum as well and will continue to keep an eye to see if anyone has had this problem. If I can find a solution I'll be sure to post all the steps I took.


The first thing I would do is to completely bypass your Netgear router. Get the service working properly with the ATA connected to your modem first. That's a good first step. If you remove the Netgear router and the problem disappears, then you know the issue involves your Netgear router. Then you can move on from there and go through the steps I listed.

Freephoneline is a bit easier to troubleshoot yourself since you can adjust settings in the ATA, whereas with Fongo Home Phone, your ATA is locked to their provisioning server. If you still have problems when directly connected to the modem, try using voip4.freephoneline.ca:6060 with your Freephoneline service. If that works then the device issued by your ISP likely has SIP ALG enabled in it somewhere, even if the setting is hidden to customers. Get SIP ALG disabled.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
User avatar
Liptonbrisk
Technical Support
 
Posts: 1840
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:8680EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin


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