HT-701 Weird problem

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HT-701 Weird problem

Postby sbougie » 02/10/2017

Hello,

I have a HT-701 with the lastest firmware, its working fine but I'm getting a weird problem, Sometime people call in, They can ear me answering the phone but I dont ear anything, but if I hangup and call them back everything is working fine

I have a Static IP for the HT-701, i tried
SIP REGISTER Contact Header Uses: WAN Address
also have Enable SIP Options Keep Alive: Yes
I have my IPS DNS written in the HT-701


And i followed the PDF instruction from viewtopic.php?f=15&t=18839

Can clue on how can I fix this problem?
Thanks
sbougie
One Hit Wonder
 
Posts: 1
Joined: 02/10/2017
SIP Device Name: HTS-701
Firmware Version: 1.0.82
ISP Name: Videotron
Computer OS: Windows 7
Router: Asus AC-3100

Re: HT-701 Weird problem

Postby Liptonbrisk » 02/11/2017

I'm not sure if a NAT hole is periodically closing for your RTP ports, some SIP ALG issue is causing problems, other devices on your LAN are occasionally hogging bandwidth, or something else. Regardless, I suggest the doing the following:


1. Ensure whatever modem/router combo your ISP gave you is in bridge mode.

2. If you can't get the router combo from your ISP in bridge mode, then either see if you can disable SIP ALG in it--or for your Primary SIP Server in your ATA, use voip4.freephoneline.ca:6060

Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall (you do already), disable whatever SIP ALG feature is enabled in the router (except for Asus routers, you shouldn't need to disable SIP Passthrough, Asus' SIP ALG setting, which does work with FPL), and stick whatever modem/router combo your ISP gives you into bridge mode. These router combos issued by ISPs frequently have faulty (and hidden) SIP ALG/SPI features enabled with no way for the customer to disable them without getting a technical representative from his or her ISP to turn this feature off. Quite frequently, the first representative you speak to will have no idea how to accomplish this, much less know what SIP ALG is. Someone may try to enable DMZ in your modem/router combo or port forward; doing either is a huge security risk. Be aware if you reset your modem or when your ISP pushes a new firmware update to your modem/router combo, SIP ALG may be enabled again by default (and, therefore, it’s simply better to have your own router with SIP ALG disabled in it).

To understand why SIP ALG often causes horrible problems, please visit http://www.voip-info.org/wiki/view/Routers+SIP+ALG.

3. Submit a support ticket requesting a "forced registration" for your account: https://support.fongo.com/hc/en-us/requests/new

4. Enable QoS in your router for your ATA. Give your ATA highest priority and all other devices on your LAN, lower priority: https://www.asus.com/support/FAQ/1008717/

5. You can try Asuswrt-Merlin third party firmware: http://asuswrt.lostrealm.ca/
Navigate to Tools-->Other settings in your router.

The following conditions should be met:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval (in your ATA) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (in your ATA)

"<" means less than

If you properly configured your ATA for FPL, NAT Keep-alive interval is 20, and SIP Registration Failure Retry Wait Time is 120.

So, after installing Merlin firmware, navigate to Tools-->Other Settings.

Change UDP Timeout: Unreplied to 10 and UDP Timeout: Assured to 100.



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, 10, 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.

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. In part, for this reason, I tend to use Asus routers. However, 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.

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 10 for UDP Unreplied Timeout and 100 for UDP Assured Timeout.



6. You shouldn't have to do the following at all with Asus routers, and port fowarding is a security risk, so only do the following if all else fails first. You could port forward from your router to your ATA, the RTP ports your ATA uses (these are UDP ports). But you shouldn't have to do that at all.

Otherwise, I have no further suggestions for you.
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
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Posts: 2763
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Router: Asuswrt-Merlin & others

Re: HT-701 Weird problem

Postby mh1983 » 02/11/2017

Lipton, do you mind if I roll your suggestions into a troubleshooting section in the 701 guide?
mh1983
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Posts: 234
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SIP Device Name: Grandstream HT-701
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ISP Name: Teksavvy Cable
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Router: Linksys WRT N600 on DD-WRT
Smartphone Model: BlackBerry Q10

Re: HT-701 Weird problem

Postby Liptonbrisk » 02/12/2017

mh1983 wrote:Lipton, do you mind if I roll your suggestions into a troubleshooting section in the 701 guide?


I don't mind.

If you're going to copy and paste stuff, maybe mention where this info is coming from (Mango, indirectly, for #5).

But, sure, I don't mind.
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: 2763
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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