Setting up Andrews and Arnold VOIP with a Grandstream HT802 v2 and Plusnet Hub One

[#1]

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@mike600 wrote:

I only know of a handful of people who have gone down the A&A route, but all with Plusnet have had issues.  Specifically with SIP ALG.

Despite it being a seemingly common issue, they haven't been able to resolve and their customer services - who are usually good - get a bit shirty when asked about it.

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@mike600 

I've been using several VoIP accounts simultaneously for over 20 years, all of that time with PlusNet, and don't recall having problems with SIP ALGs (mostly disabled), always using DrayTek VoIP routers until about 2.5 years ago when I switched to a non-VoIP DrayTek.  For a number of years I also used a Gigaset N300 DECT VoIP device (again with several simultaneous accounts) alongside the multiple VoIP connections of the DrayTek routers, but at present I'm only using the Gigaset device.

Almost a year ago I ported my BT landline number to A&A and everything has been fine since Day1, i.e. no issues experienced whatsoever.

That said, I'm not claiming to be a VoIP expert, just a regular user with no problems to speak of.

[#2]

@mike600 I'd concur with @RobPN , I've used voip for a considerable number of years both at home on a Plusnet connection and in the office on Zen & BT connections. The only router to give problems in my experience the BT HH5 & Plusnet Hub 1 due to the fact that SIP ALG cannot be disabled by the user. In my experience almost all implementation of SIP ALG are broken and they should be disabled.

Basically being behind a NAT router 'breaks' SIP and so 'tweaks' have to be implemented to make it work. In the early days of SIP takeup, this meant using STUN or outbound proxies. Nowadays, most SIP providers implementations are NAT aware and usually all that needs doing by the user is to enable 'keep alive' to keep the NAT pinhole open on their router.   

[#3]

I have seen this death spiral with SIP/ALG so many times.

Once you step away from SIP/ALG port range for both SIP & RTP, its effect is gone and you should see your devices connected on the configured ports in the VoIP via providers management portal.

If the VoIP phone still does not work as expected some other network or config issues needs to be resolved,

Things like CGNAT,  ISP Firewalls,  Software Firewalls, Double NAT, Codec choice try G.711a are well capable of affecting in bound voice.

Not Ringing can be ATA ringer voltage too low / frequency not 25 Hz / shape / cadence too short try (60)2/2 

[#4]

Following up what @RobPN wrote, I've also got a Gigaset VoIP "base station" and Plusnet.

I don't use a Plusnet router, instead using a Draytek which is capable of what it calls "VoIP Prioritization".

As I've a 145 Mb/s full fibre connection, I suspect this isn't of great importance but with a more limited connection I think that would help.

Hence my earlier suggestion of replacing the Plusnet device.

[#5]

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@daniellam wrote:

Interesting you mention the QoS setting... isn't this going to affect many more PN customers as we get nearer to the PSTN switch-off in January 2027, if they decide they want a VOIP service whilst keeping their broadband with PN? Unfortunately, there isn't a "Hub Three" that is 100% VOIP ready...

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Yes, if it's a slow connection. Trying to use the phone whilst little Johnny is downloading or wanting priority whilst playing his latest game is likely to cause problems of buffering/stuttering to phone calls. Voice needs very little bandwidth, but it can't tolerate interruptions caused by other traffic. You can only use the available bandwidth once. 

Routers with a built in ATA such as those supplied by BT/EE have no user settable QoS, but by design it is already incorporated into the voice channel.

Plusnet do not offer a VoIP service so have no reason to offer a VoIP ready router. Add to that, BT/EE use their own voice data protocol so that they can't be used with other providers.

[#6]

I'm another A&A VOIP user on SOGEA with a PlusNet Hub 2 and have had no problems or complications whatsoever.

Equipment is Gigaset N510 (previously N300 during the POTS to VOIP transition) and now Gigaset handsets though I used an assortment of old DECT ones initially.

[#7]

We started with the Cisco ATA and dual VoIP accounts and they just worked on Plusnet ADSL.
When we changed ISP to Virgin Media the Cisco ATA just worked again, zero changes needed.

I read over and over in this and similar posts about SIP/ALG and I can only conclude that people have either chosen to lock horns with a known problems specific Hubs and / or lack a detailed grasp of what SIP/ALG does and how to side step it.


One way voice can also be caused by
+ Incomplete / Incorrect configuration of VoIP equipment
+ Firewalls
+ CGNAT
+ Double NAT
+ VPNs
+ Ad-Blockers ( DNS or URL )
+ Use of an unsupported Codec

Failing to ring reliably can be caused by the issues above and ;
+ Ring Voltage not high enough
+ Ring Frequency not 25Hz
+ A phone that does not respond well to Sine or to Triangle waves
+ Ring Cadence has too short an on period & looks to the phone like a dropped call
+ Line impedance not to UK style

Whilst this thread remains in the its all a SIP/ALG fault, the deadlock will continue.

[#8]

Hi @PhilipHeyes - thanks for the continued info. Interestingly, yesterday I had an inbound caller and was able to hear them first time around - even though I hadn't made the SIP port change yet. I have done so now (moved it to 15060), so let's see if that is able to sidestep the SIP ALG.

On incoming ringing, that's never been a problem. I have a BT Viscount (9515AR) connected (as well as a pair of Gigaset A420's) and the Viscount always rings, given I've set the "Ring Frequency" to 25Hz and the "Ring Power" to 50Vrms on the Grandstream HT802 v2. Also, I've reused an old microfilter that contains a capacitor to emulate the third "bell" wire and it works just fine.

Codec was already set to PCMA (G.711 A-law), as it's the only one supported by A&A.