why the proxy server change?

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why the proxy server change?

Postby maximebeauvais » 03/02/2020

Hello,

Over the last year, every 2-3 months, I have to change the proxy server name because freephoneline stop to work.

The name that I have to enter is voip.freephoneline.ca OR voip2.freephoneline.ca

I always have to alternate. is it normal?

Thanks

PS: my ATA is an OBi100
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maximebeauvais
Active Poster
 
Posts: 77
Joined: 09/23/2014
SIP Device Name: OBi110
ISP Name: Bell Fibre 25/25
Router: Bell Home Hub 2000

Re: why the proxy server change?

Postby Liptonbrisk » 03/02/2020

maximebeauvais wrote:
Over the last year, every 2-3 months, I have to change the proxy server name because freephoneline stop to work.


1) What happens? Do you not hear a dial tone? Does your ATA become unregistered?

2) When this issue occurs, what does SIP Status indicate after logging in at https://www.freephoneline.ca/showSipSettings ?
Also check to ensure that SIP User agent shows a device you recognize when this problem occurs.


Registration or a "connected" status is required for incoming calls. Registration is not required for outgoing calls.
Note that only one registration per FPL account is allowed at any time. When there are multiple devices/softphones using the same account, only the most recent registration is valid. The previous device will lose registration and incoming calls will not work on it.


3) The next time this problem happens, dial ***1, and enter the IP address you're told into your web browser.

If you used the Obitalk web portal (http://www.obitalk.com) to configure your ATA, keep in mind that you must continue using it to configure your ATA. Otherwise whatever settings you change will eventually be overwritten by what you previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1. Enter that IP address into a web browser. Navigate to System Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA. Now obitalk.com won't overwrite whatever changes you make via the device's interface (via IP address).

Pick one method (obitalk.com) or the other (IP address of device) for changing device settings. Never use both methods.

If you're using Obitalk.com, look at this picture for reference: http://i.imgur.com/xwyqgRE.jpg
That grey cog wheel with the "E" is for the expert configuration menu.

It shows when logging in at http://www.obitalk.com, selecting "Edit Profile" on the left, then scrolling down
under "Advanced Options" and finally selecting "Enable OBi Expert Entry from Dashboard."

If you're using Obitalk.com, you will need to enter the expert menu.

i) navigate to Voice Services-->SP(FPL) Service-->X_UserAgentPort. X_UserAgentPort should be a random port number between 30000 and 60000. Just pick a port number in that range. Change to a new port number in that range. Click the “submit” button, and reboot the ATA. (If you use Obitalk.com to change settings, you will need to use Obitalk.com).

If changing X_UserAgentPort fixes the problem, you were dealing with a corrupted NAT connection in your router.
Possibly a NAT router connection was never disconnected or never timed out properly. And, then, the
ATA keeps the corrupted connection in a persistent state over and over again. (Credit goes to Mango for
this information). Possibly, this problem is due to the router's UDP timeout being in excess of the ATA's
Failure Retry timer (RegisterRetryInterval with Obihai ATAs). With FPL, that's 120 seconds.

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) < 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, 17, 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_KeepaliveExpires is supposed to be 20 with FPL.
Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin, third party firmware for Asus routers, does offer
easy access to these two settings, which are found under Tools-->Other settings. However, my understanding is that third party Tomato firmware has these two settings as well. Mikrotik routers do as
well: https://forums.redflagdeals.com/recomme ... #p28056619. 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. The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 for UDP Unreplied Timeout and 117 for UDP Assured Timeout.


is it normal


No. The only time you should have to switch ProxyServer is if you're temporarily IP banned due to too many registration attempts within a short period.

Keep the following in mind:
https://community.freepbx.org/t/trunk-s ... ca/22479/7
"As May 2013, our servers will rate limit REGISTER requests to a maximum of 10 requests per 5 minutes. Each authentication round usually consumes 2 requests (digest auth), so it is a fair number given our guidelines. Also, it does not affect INVITES (which are also authenticated)…

This rate limit is applied per IP address as our service is tailored to residential Canadian users (ADSL/Cable)."

So, it's important that after navigating in your ATA to ITSP Profile (FPL)-->SIP-->
a) RegistrationPeriod is set to 3600 seconds, and
b) RegisterRetryInterval is at least 120 seconds
https://support.freephoneline.ca/hc/en- ... redentials

Each time you reboot, your ATA is attempting to register again. So be careful about how many registration attempts you're trying within 5 minutes.
By the way, proper device reboot order is always modem-->router (wait for it to be fully up and running first)-->ATA.
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: 2764
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: why the proxy server change?

Postby Liptonbrisk » 03/02/2020

Also . . .


Typically, for VoIP SIP services, especially for freephoneline, you want

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

Visit https://www.think-like-a-computer.com/2 ... es-of-nat/. 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) < SIP OPTIONS Keep Alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires) < 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, 17, 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 ... 2115672/2/

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 for UDP Unreplied Timeout and 117 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: 2764
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others


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