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UHD

499 posts

Ultimate Geek
+1 received by user: 206


  Reply # 1480624 28-Jan-2016 12:41
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Skinny, you guys really need a low user option. Something in the 10-25GB per month range.


322 posts

Ultimate Geek


  Reply # 1481601 30-Jan-2016 08:12
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My VoIP calls get dropped sometimes because VoIP line gets disconnected. Then my phone tries to re-register to the VoIP server. It never happened to me on a fixed line modem. Might be a issue with skinny CG NAT? 4G signal is 3 bar, -97 to -100. Incoming and outgoing calls work fine, but its just that some calls get dropped because the line is broken.

 
 
 
 


354 posts

Ultimate Geek
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  Reply # 1481653 30-Jan-2016 10:37
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As per UHD's comment, a low use plan would ideal for the folks holiday home as they can afford but can not justify a fixed line solution for a property they only visit four days a month max, maybe a week and a half at christmas (retirement at 60 is a pipe dream to most), but a competitivly priced 10gb option would be ok as they do like to stream content, albeit one or two tv shows a day.

626 posts

Ultimate Geek
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  Reply # 1481714 30-Jan-2016 12:48
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What would service in a 'fair' coverage area be like typically? I guess 10-20mbit?


BDFL - Memuneh
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  Reply # 1481730 30-Jan-2016 13:31
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See my previous comment. 20Mbps on two bars (Martinborough), 40Mbps on four bars (Johnsonville, Whitby).

 

YMMV.





322 posts

Ultimate Geek


  Reply # 1481889 30-Jan-2016 17:33
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Ping is around 20 to 25. Signal 3 bar 45down/12up max. Signal 2 bar 25down/6up max. Not bad. Quite nice.
But online gaming suffers. I think 4g speed is still quite good when signal drops to 2 or 1 bar, compared to 3G, you know

322 posts

Ultimate Geek


  Reply # 1481894 30-Jan-2016 17:40
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When 4G was first available, I could get 96mb down. But now, I can get only half that. I think its because too many people are on 4G now.
Don't know what the speed will become when many people get on skinny broadband and watch YouTube at the same time, the speed might suffer a lot. Don't know if 4G cellsite has any fiber backup or not.
And besides, how can a 4G cellsite handle so many skinny broadband users watching YouTube at the same time?? Won't the cellsite get overloaded ??

68 posts

Master Geek
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  Reply # 1482185 31-Jan-2016 12:08
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40 down 20 up, 4 bars here in little ol' Hoki

 

 

 

I can live with that...laughing


QSX

8 posts

Wannabe Geek
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  Reply # 1482993 1-Feb-2016 17:24
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Hi All,

 

Has anyone else managed to maintain VoIP for incoming calls longer than a day or two?

 

If you have had success in this area I would be interested in how you have configured VoIP. The only combination I have not tried is using STUN with SiP ALG, mainly  because I have never used STUN in the past even though I have used 4G accounts behind CGNAT with different overseas carriers for over 12 months (for each carrier) without issue. I am away from home at present so I cannot change the configuration (only test the one I have by calling it).

 

I have two SIP providers which I have tried independently (i.e. only one connected to the router at a time, not both) and both maintain registration 27/7 such that I can always make outgoing calls (audio both ways) using the SIP ALG. However after 1 to 2 day's max (usually about 24 hours) I am unable to receive incoming calls (i.e. the phone will not ring and call goes to the answering phone service at the SIP server). Hence you would never know looking at the ATA or making a test call using the VoIP phone that incoming calls have failed (because the ATA is still registered and outgoing calls work fine).

 

When this occurs, as I explained in my previous post, but will state again for clarity, changing the router to place the ATA into the DMZ zone and switching off SIP ALG allows the incoming call to ring the phone, but the incoming RTP gets lost (i.e. no incoming voice, only outgoing voice). Switching off the firewall and/or attempting to use STUN does not solve this issue. Rebooting the ATA does not solve this issue.

 

Rebooting the router with the SIP ALG enabled (and DMZ disabled) allows the router to establish full functionality until the iP address changes. I suspect the SIP ALG does not update. Disabling SIP ALG does not solve the issue because of the way the router handles IP address in DMZ etc. which does not allow the ATA to pick up the correct IP addresses etc. to establish RTP.

 

I have another 4G router setup (totally different to the B315) that I have placed a Skinny mobile SIM into on 4G and opened up the VoIP port ranges (even though I am behind CGNAT, it allows the ATA to establish correct communications). I have had this running for about 3 days now and it continues to work as expected. I expect the 4G data to work in the same way as Skinny broadband hence the only difference is the router. Thus I conclude that the problem lies with the B315 router in my opinion, in that it must handle IP address differently and not allow the ATA to determine the correct external IP address to route the call without the SIP ALG (a casual look at the SIP logs also suggests this is correct, and there are no configuration settings I have tried on the ATA that will correct this to working state, including STUN settings with a STUN server). If I am correct I expect all VoIP users to experience the same issues with incoming calls using the B315 router. If you are not then I would be very interested to know what you are doing differently (assuming you are using SIP and not IAX or Skype etc).

 

QSX


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  Reply # 1483008 1-Feb-2016 17:48
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I used a Cisco SPA112 auto provisioned with 2talk. I haven't done any long phone calls, but other testing has been fine. It's been on solid probably a couple of weeks and I can still call it anytime and it rings and I can answer without fail.

 

 





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QSX

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Wannabe Geek
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  Reply # 1483216 2-Feb-2016 07:55
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coffeebaron:

 

I used a Cisco SPA112 auto provisioned with 2talk. I haven't done any long phone calls, but other testing has been fine. It's been on solid probably a couple of weeks and I can still call it anytime and it rings and I can answer without fail.

 

 

Thanks for your reply. The only difference I can see looking at 2talks SPA122 setup is the use of STUN server (optional) and an aggressive retry and long expiry interval. The SPA122 is basically the same as a SPA112 but also has a Ethernet port and DHCP router. 

 

I looked at 2talks SPA122 page a week ago and that is why I commented that the only thing I have not tried is using STUN with the SIP ALG. 2talk state this is optional and I have never needed to use STUN in the past to make a reliable system, however I will try it with the SIP ALG (I did try it without SIP ALG to try and get rid of the SIP ALG but not suprisingly I had no success). I don't believe the short retry and long expiry intervals will have any impact because restarting the ATA does not fix the issue.


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Biddle Corp
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  Reply # 1483218 2-Feb-2016 08:01
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VoIP on Spark/XT has never worked well historically and there are many threads on here discussing this. It's probably not surprising it still doen't work well.

 

Port forwards in the VoIP world should not be used and post a significant security risk. If they are needed a setup is broken, so you need to fix the setup. If your house door breaks you fix it, you don't leave your door wide open as your means of getting in.

 

A SIP registration is typically for inbound calls, not outbound (it's basically just saying "here's my IP to send calls to"). 99% of VoIP providers will allow an outbound call regardless of the registration state as your user credentials are sent in the call setup. You can quite easily have a registration that's still up but inbound calls fail particularly in the CG-NAT/double NAT world.

 

 

 

 


QSX

8 posts

Wannabe Geek
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  Reply # 1483813 2-Feb-2016 19:01
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sbiddle:

 

VoIP on Spark/XT has never worked well historically and there are many threads on here discussing this. It's probably not surprising it still doen't work well.

 

Port forwards in the VoIP world should not be used and post a significant security risk. If they are needed a setup is broken, so you need to fix the setup. If your house door breaks you fix it, you don't leave your door wide open as your means of getting in.

 

A SIP registration is typically for inbound calls, not outbound (it's basically just saying "here's my IP to send calls to"). 99% of VoIP providers will allow an outbound call regardless of the registration state as your user credentials are sent in the call setup. You can quite easily have a registration that's still up but inbound calls fail particularly in the CG-NAT/double NAT world.

 

 

 

 

 

Thanks for the information, and clarifying SIP registration. I did not realise that there was a historic issue with VoIP on Spark/XT (its hard to be across every post on a forum, especially when one is new and not reading the forum every day). 

 

QSX


322 posts

Ultimate Geek


  Reply # 1483852 2-Feb-2016 20:02
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QSX:

Hi All,


Has anyone else managed to maintain VoIP for incoming calls longer than a day or two?


If you have had success in this area I would be interested in how you have configured VoIP. The only combination I have not tried is using STUN with SiP ALG, mainly  because I have never used STUN in the past even though I have used 4G accounts behind CGNAT with different overseas carriers for over 12 months (for each carrier) without issue. I am away from home at present so I cannot change the configuration (only test the one I have by calling it).


I have two SIP providers which I have tried independently (i.e. only one connected to the router at a time, not both) and both maintain registration 27/7 such that I can always make outgoing calls (audio both ways) using the SIP ALG. However after 1 to 2 day's max (usually about 24 hours) I am unable to receive incoming calls (i.e. the phone will not ring and call goes to the answering phone service at the SIP server). Hence you would never know looking at the ATA or making a test call using the VoIP phone that incoming calls have failed (because the ATA is still registered and outgoing calls work fine).


When this occurs, as I explained in my previous post, but will state again for clarity, changing the router to place the ATA into the DMZ zone and switching off SIP ALG allows the incoming call to ring the phone, but the incoming RTP gets lost (i.e. no incoming voice, only outgoing voice). Switching off the firewall and/or attempting to use STUN does not solve this issue. Rebooting the ATA does not solve this issue.


Rebooting the router with the SIP ALG enabled (and DMZ disabled) allows the router to establish full functionality until the iP address changes. I suspect the SIP ALG does not update. Disabling SIP ALG does not solve the issue because of the way the router handles IP address in DMZ etc. which does not allow the ATA to pick up the correct IP addresses etc. to establish RTP.


I have another 4G router setup (totally different to the B315) that I have placed a Skinny mobile SIM into on 4G and opened up the VoIP port ranges (even though I am behind CGNAT, it allows the ATA to establish correct communications). I have had this running for about 3 days now and it continues to work as expected. I expect the 4G data to work in the same way as Skinny broadband hence the only difference is the router. Thus I conclude that the problem lies with the B315 router in my opinion, in that it must handle IP address differently and not allow the ATA to determine the correct external IP address to route the call without the SIP ALG (a casual look at the SIP logs also suggests this is correct, and there are no configuration settings I have tried on the ATA that will correct this to working state, including STUN settings with a STUN server). If I am correct I expect all VoIP users to experience the same issues with incoming calls using the B315 router. If you are not then I would be very interested to know what you are doing differently (assuming you are using SIP and not IAX or Skype etc).


QSX


Lol. I found a easy fix for me last week. Add another 0 to port 5060, lol, so easy. My iTalk works well now, incoming calls OK for a few days, no problems. I don't know about my 2talk , its registered on the modem, but I don't use it, so I don't know if it works or not.
Or you can try 5060 --- 5070, or whatever voip port. And Please post the results here. I tried everything else, but nothing works, except this method.

QSX

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Wannabe Geek
+1 received by user: 3


  Reply # 1484181 3-Feb-2016 10:07
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I found a easy fix for me last week. Add another 0 to port 5060, lol, so easy. My iTalk works well now, incoming calls OK for a few days, no problems. I don't know about my 2talk , its registered on the modem, but I don't use it, so I don't know if it works or not.
Or you can try 5060 --- 5070, or whatever voip port. And Please post the results here. I tried everything else, but nothing works, except this method.


Interesting. 2talk allow registration on 50600 which would place it above the SIP ALG. I was wondering this morning if that would work. I suspect changing the SIP AGL port may acheve the same result. I will try it soon. Thanks for sharing.

QSX

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