FTTP and digital voice - how does bandwidth allocation work

Hoping an ISP or Openreach (OR) person can advise me on this.

Have Zen FTTP delivered over OR fibre (installed 2018) and upgraded to 300 Mbps down and 47 up in Feb on a retention deal.

Am using one of their Fritzbox 7530 routers and am preparing to ditch my landline and switch to digital voice (mobile reception is abysmal here).

Also had some issues with a phone system at work which made me think (Zen support have been great but can't find anything, I've collated log files and am waiting to hear back from colleagues who spend their day managing that particular system) however it got me thinking....


I'm currently weighing up whether to go with Zen for digital voice or Andrews and Arnold (ran a quick test with them back in 2020)

Now I'm assuming if you go with a single supplier they slightly raise the upstream and downstream on OR so that even if the connection is maxed out DV will still work and if I were to stick with Zen for the connection and go with A&A for their voice service this wouldn't happen or can I request a modification to accommodate this?

A&A would be cheaper on the basis that the phone isn't used all that much.

Hope this is clear and thanks for help in advance.

No CP, Zen or any other is able to raise the speeds to add a bit extra for VOIP, it can't be done that way. The correct way is to use QOS (Quality of Service) settings on the router to reserve a small amount of prioritised bandwidth for VOIP traffic, the speeds required for VOIP are quite low.

The question is do CP supplied routers support this and is something that will be set up by default? I don't know, my router is an enterprise grade device and various protocols can be configured with different priority levels and bandwidths if required.

You need to ask Zen how they give priority to VOIP traffic.

If it helps, I've been using A&A VOIP over my 80/20 FTTC connection without any issues or any QoS rules set, despite lots of other network activity.

On a 330/50 service you'll be just fine.

In reply to a post by techguy:

I'm assuming if you go with a single supplier they slightly raise the upstream and downstream on OR so that even if the connection is maxed out DV will still work

As the digital voice is not on its own Vlan it wouldn't help, if there was a way of adding another 128Kbps required for the digital voice that would still get saturated if you try to max out the broadband service.

In reply to a post by ferretuk:

If it helps, I've been using A&A VOIP over my 80/20 FTTC connection without any issues or any QoS rules set, despite lots of other network activity.

On a 330/50 service you'll be just fine.

That's reassuring to know, thanks.

techguy

You need to ask the ISP.

There are several ways to provide VOIP.

1. Over the top VOIP. Just VOIP in with everything else.
2. As above but with QOS enabled to prioritise the VOIP packets.
3. Set up on a separate VLAN (Takes approx. 128Kb from Up and Down capacity) BT's implementation is like this.

The ATA (CODEC) also make a difference as best quality voice is very small packets (10 -20ms packets) but small packets decrease the available Bandwidth as you do not fill the available frame and increase the router load. This should not be an issue with a reputable ISP router.

Look up G711 / G 729,/G723 etc.

Lower quality VOIP would use bigger packets
if using 2. you need to make sure the VOIP is the highest priority.

Unless you are really hammering your connection all three work just fine most of the time. I would expect any Router with a dedicated voice port to use 2 or 3. 3rd party VOIP often uses 1 but works just fine for most people with Speeds of over 10Mb upstream. Speeds under 10Mb upstream may occasionally struggle using 1...

The Fritz!Box has excellent QoS built-in, and defaults VoIP to the realtime class. I have mine configured with A&A and it works beautifully. It would work just as well with Zen’s voice service.

I added three Gigaset DECT phones and paired them directly to the Fritz!Box. I configured the Fritz!Box to sync my contacts from iCloud so the contacts on the phones match my iPhone / Mac.

In terms of mobile coverage, I always select a mobile service that supports wifi calling and text (Vodafone, BT Mobile, T-Mobile and many more). If the phone supports it, calls and texts run over wifi instead of the mobile data network. Dead handy in places with wifi but no mobile signal; like my recent holiday home in the Lake District. Could be useful if you have poor coverage at home.

In reply to a post by techguy:

In reply to a post by ferretuk:

If it helps, I've been using A&A VOIP over my 80/20 FTTC connection without any issues or any QoS rules set, despite lots of other network activity.

On a 330/50 service you'll be just fine.

That's reassuring to know, thanks.

I use a VoIP system and even on my 36ish Mb/s FTTC broadband, I don't have any problem and that is also when the TV is streaming 4K.

I use Sipgate as my provider and a very old Linksys PAP2, you would have to really bang your internet connection to make a difference to VoiP.

If you were in a corporate with dozens or hundreds of simultaneous voice channels then you would be best placed to have them on a a separate VLAN and running QoS on all your routers and switches.

However for a domestic or small/home office internet connection with one or perhaps a handful of simultaneous VoIP channels there really isn’t much wrong with running VoIP plainly over the top without any dedicated bandwidth reservation or QoS. It will be fine.

I’ve run VoIP over a variety of broadband connections and terrestrial connections were never a concern. 4G/5G connections are generally OK too, however they are far more liable to delay and jitter variability - which will impact call quality - however that’s not a concern for you.

I hate to chip in amongst such illustrious company but I'm sure I recall being told many years ago that the frequency response of a voice telephone line was (iirc) 400Hz to 4KHz. A decent hifi system will provide a frequency response around 20Hz to 20KHz so on a line that is running at Megabit speeds it's really not something that you're going to notice.

Sorry but the analogue frequencies of telephone calls are not directly related to the digital transmission of data by VOIP. So your analogy is incorrect.

I vaguely recall, from about a million years ago, that the digital sample rate of an analogue source needs to be twice the frequency in order to record it accurately. See Nyquist's theorem. So if the maximum frequency is (say) 4.4KHz them a sampling rate of 8.8Kbps is needed. Add in a bit extra and round it up for better sound quality and 10Kbps ought to do the trick.

Edited by JHo1 (Sat 27-Aug-22 11:24:47)

In reply to a post by Realalemadrid:

Sorry but the analogue frequencies of telephone calls are not directly related to the digital transmission of data by VOIP. So your analogy is incorrect.

Why is he wrong?

The wider the frequency range, the greater the data to be sent, analogue or digital and as such the more bandwidth in this case would be required.

The analogue signal is converted to digital and then sent, exactly the same applies to music and video too. Granted you then start to get into compression ratios and codecs etc however generally the more you put in, the bigger the output, unless you know of some magic that changes this?

In reply to a post by JHo1:

I vaguely recall, from about a million years ago, that the digital sample rate of an analogue source needs to be twice the frequency in order to record it accurately. See Nyquist's theorem. So if the maximum frequency is (say) 4.4KHz them a sampling rate of 8.8Kbps is needed. Add in a bit extra and round it up for better sound quality and 10Kbps ought to do the trick.

You were doing well up until the last sentence

Each sample will require a number of bits to indicate its magnitude, usually a minimum of 8 for reasonable quality, so a sample rate of 10khz needs a bit rate of at least 80kbps.

Higher sampling rates are better but not necessary. A CD of uncompressed music uses 44.1kHz though it may have been down sampled from a higher frequency recording. You're right though in that our sample is not a single bit but rather, well how many? I think we may be furiously agreeing here.

Either way I think we can agree that, even without compression, we're talking in the low kbit range and the impact of a voip conversation on a domestic multi megabit line is going to be very hard to notice.

I found this https://www.nextiva.com/blog/voip-codecs.html which describes three voip codecs, two at 64kbps and one at.... eight! I'm sure there are others

Edited to add nextiva website

Edited by JHo1 (Sat 27-Aug-22 12:22:38)

I have VoIP in its own queue on pfSense so can monitor exactly how much the bit rate is.

Using codec G711, the codec encodes at 64Kbps and the data rate seen in pfSense queue is around 92Kbps. This is a bit higher than what is usually shown online for total expected rates for this codec when overheads are added, which I think in my case is because SIP packets are also included in that queue inflating it slightly. This is in one direction, so it is 92Kbps down and 92Kbps up. With other codecs common to VoIP it can be considerably less than this, for example G729 is around 27Kbps with overheads.

One of the issues with VoIP is the packets are quite small, so the transfer of audio packets are fairly inefficient, but not usually a problem in the home for a couple of calls. A large call center using VoIP would need pretty fast kit to cope with the high number of packets it has to process.

Overall VoIP is very low bandwidth, and given our super fast connections in comparison it is fairly hard to completely max the line out so much that VoIP packets can not find a way through.

The typical codecs used in VoIP are not new. There are better sounding codecs for the same or lesser data rate. For example G711 is from 1972 but still widely used today, one reason it is still used is that the patent has expired so it is free, and pretty much supported by every bit of kit.

Edited by E300 (Sat 27-Aug-22 13:45:43)

Either way I think we can agree that, even without compression, we're talking in the low kbit range and the impact of a voip conversation on a domestic multi megabit line is going to be very hard to notice.

I found this https://www.nextiva.com/blog/voip-codecs.html which describes three voip codecs, two at 64kbps and one at.... eight! I'm sure there are others

You’re correct in that the total bandwidth for a single voice channel is quite small, but the perceived “impact” from the OP question is not the actual consumed bandwidth but the impact other traffic may have on the real time stream - that is would it be affected by other traffic that could delay the sending or receiving of the packets for the voice connection - factors we know as round trip delay (ping) and jitter. The evidence is that unless a given broadband connection is very, very highly used then in reality for a residential connection with one or perhaps a handful of simultaneous voice connections it will generally be OK. So although you could mitigate by partitioning the available data path (virtual LAN) or by prioritising voice traffic by tagging the packet (QoS) it’s not really necessary on a domestic connection, or at least it’s positive impact is probably quite negligible.

On the other hand corporate connections with much higher volumes of VoIP (simultaneous channels) are a different matter and it is good practise by the service provider on something like a leased line WAN connection to partition some of the available bandwidth into a separate VLAN for VoIP/Telephony so that it is not “crowded out” by other less delay sensitive traffic on the main VLAN.

The sound quality of voice connection is typically measured by something called a Mean Opinion Score (MOS). Have a search for it and do some extra background reading whilst you are looking at VoIP codecs. There is a lot of history (and some corporate politics) with certain wideband codecs especially and there has been a lot of evolution in the last few years.

Most standard VoIP interconnections have for decades followed the TDM world of telephony and have defaulted to G.711 a or u-law codecs which are a pretty good trade off for quality vs DSP processing requirements vs bandwidth (the three conflicting holy grails of voice codecs).

Here is some pretty good starter for ten, background on codecs:

https://info.teledynamics.com/blog/demystifying-code...

https://info.teledynamics.com/blog/the-wonderful-wor...

The most common VoIP codecs you’re going to find are:
G.711 A-law
G.712 u-law
G.722
Opus
iLBC
G.729A
GSM

Some of the best quality calls will be proprietary platform such as Apple FaceTime calls which use I believe use the AAC-ELD codec for the voice element of calls.

But is 8 bit actually needed for voice? Then throw in the codec and compression.

We managed to get reasonable voice quality, that was acceptable across 2.4 and 4.8 kbps links although most was at either 4.8 or 9.6 kbps!

This is all interesting in a nerdy way

Apple FaceTime is amazing, when came out of hospital a few years ago I stayed with my sister and brother-in-law for a week and they used an Ipad to chat to my nephew, even on a naff ADSL connection the quality was excellent, both video and audio.
Far better than anything I saw on Skype or other systems at the time.

Thanks for all the responses, most informative and interesting.

I'll probably go with A&A when the time comes as it didn't get used an awful lot and (at the time of writing) £1.20 per month to host a number is better than £7.

In reply to a post by techguy:

Thanks for all the responses, most informative and interesting.

I'll probably go with A&A when the time comes as it didn't get used an awful lot and (at the time of writing) £1.20 per month to host a number is better than £7.

i use sipgate as I said, costs nothing unless I use it.

i use sipgate as I said, costs nothing unless I use it.

Yes, I have petrol in my car that works like that. 🤔

In reply to a post by Zarjaz:

Yes, I have petrol in my car that works like that. 🤔

Don't forget car insurance as well, that comes at a cost if you use the car or not

In reply to a post by zyborg47:

i use sipgate as I said, costs nothing unless I use it.

I always worry that companies with thousands of users earning £0 won't be there when I need them. The costs of running the infrastructure aren't £0.

Yes, I'd rather pay a modest monthly charge and hope that means my phone nbr is less likely to be lost somehow.

In reply to a post by jchamier:

In reply to a post by zyborg47:

i use sipgate as I said, costs nothing unless I use it.

I always worry that companies with thousands of users earning £0 won't be there when I need them. The costs of running the infrastructure aren't £0.

i think there are enough people using sipgate to keep it going. I have a tenner on it and that tenner have been there for a couple of years, I really don't use it that much. To be honest, the only reason I still have it is that I give my work the number as I don't want them to have my mobile number and my brother phone me on it for some reason

In reply to a post by dect:

In reply to a post by Zarjaz:

Yes, I have petrol in my car that works like that. 🤔

Don't forget car insurance as well, that comes at a cost if you use the car or not

Yes, but my point was a gentle poke at Adrian’s post …. a bum analogy really, but hopefully the drift was caught.

In reply to a post by zyborg47:

In reply to a post by techguy:

Thanks for all the responses, most informative and interesting.

I'll probably go with A&A when the time comes as it didn't get used an awful lot and (at the time of writing) £1.20 per month to host a number is better than £7.

i use sipgate as I said, costs nothing unless I use it.

I did a quick test with sipgate too but found the CLI a little hit and miss at the time I tested it and a traceroute went out to Germany (I back my PC up to a service in the US so location doesn't bother me) but I'll stick with a company whose infrastructure is in the UK for this purpose I think.

In reply to a post by jchamier:

In reply to a post by zyborg47:

i use sipgate as I said, costs nothing unless I use it.

I always worry that companies with thousands of users earning £0 won't be there when I need them. The costs of running the infrastructure aren't £0.

Me too, also, self service and backend automation is great but sometimes you need to consult someone who knows how to recify problems when the automation breaks.

Full disclosure: I've worked for nearly 20 years in Software Support/Service Desk roles so that may colour my view on this somewhat

In reply to a post by Zarjaz:

In reply to a post by dect:

In reply to a post by Zarjaz:

Yes, I have petrol in my car that works like that. 🤔

Don't forget car insurance as well, that comes at a cost if you use the car or not

Yes, but my point was a gentle poke at Adrian’s post …. a bum analogy really, but hopefully the drift was caught.

Made me laugh.

In reply to a post by Zarjaz:

In reply to a post by dect:

In reply to a post by Zarjaz:

Yes, I have petrol in my car that works like that. 🤔

Don't forget car insurance as well, that comes at a cost if you use the car or not

Yes, but my point was a gentle poke at Adrian’s post …. a bum analogy really, but hopefully the drift was caught.

Leaving the rest of us to paper over the cracks . . .

In reply to a post by jchamier:

In reply to a post by zyborg47:

i use sipgate as I said, costs nothing unless I use it.

I always worry that companies with thousands of users earning £0 won't be there when I need them. The costs of running the infrastructure aren't £0.

That's why I went with AAISP too, then found it has a super useful feature to automatically e-mail a copy of every call to you (as both of us in the household have awful memories).

Also the support, I found the COVID-19 helpline didn't work, contacted them and it was fixed really quickly.

We both have mobiles on PAYG which inevitably run out of credit at the worst moments, so having VoIP use top-ups too was a no go. Don't want it running out of credit after you've just spent an hour waiting on hold.

A&A have the reputation of the Rolls Royce in VOIP, but at one time they refused to transfer numbers out. Sipgate have been around so long that their modus operandii works well and I have never had a problem with them, so would recommend for lolw volume users.