VOIP…is it worth the aggro?

[Spike]

A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.

These include but are not limited to:

- finding a suitable VOIP provider

- finding suitable equipment, such as an ATA or modem/router with it built
in (uncommon)

- setting up the equipment with the correct information

- ‘tweaking’ the equipment settings

- ‘tweaking’ the supplier’s settings

- much testing to ensure it actually works properly

- finding that in some cases some settings aren’t tweakable.

One forms the opinion that the exercise bears some similarity to banging
one’s head against a wall, always presuming you can find the right sort to
bang against.

In the presence of a well-developed mobile phone system, which for the vast
majority of people ‘just works’, why bother with VOIP? It seems like a
futile and nerdy exercise in clinging on to something that is well past its
time and which doesn’t translate well into the 21st century.

YMMV

[Marco Moock]

Post by Spike
A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.
- finding a suitable VOIP provider

Indeed that is sometimes a problem because some don't support IPv6 and
that means the service can't be used when no public routed IPv4 is
available.

Post by Spike
- finding suitable equipment, such as an ATA or modem/router with it
built in (uncommon)

Many devices are available that combine modem, router, WiFi and ATA,
e.g. from the manufacturer AVM for home users.

Post by Spike
- setting up the equipment with the correct information

Instructions are available and if you need help, ask here.

Post by Spike
- ‘tweaking’ the equipment settings
- ‘tweaking’ the supplier’s settings

I never needed to do that.

Post by Spike
- much testing to ensure it actually works properly

Rather easy.
In Germany analog lines only exist for analog only contracts and the
vast majority of fixed-line customers uses VoIP anyway. No doomsday,
even when that was predicted.

Post by Spike
In the presence of a well-developed mobile phone system, which for
the vast majority of people ‘just works’, why bother with VOIP? It
seems like a futile and nerdy exercise in clinging on to something
that is well past its time and which doesn’t translate well into the
21st century.

VoIP simply works and is being used by millions of people worldwide.
The problems from the beginning are mostly gone.
I experienced that the latency in VoIP is smaller than via mobile
cellular network, so I prefer VoIP.

[Brian Gregory]

I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and keep
it alive.

[David Higton]

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and keep
it alive.

It's easier with IPv6 because no NAT is involved.

David

[Brian Gregory]

Post by David Higton

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and keep
it alive.

It's easier with IPv6 because no NAT is involved.

Not really. If your VOIP provider knows their stuff they will set things
up so that nothing needs to be specially configured on the router no
matter whether you use IPv4 or IPv6, except, if required, QoS to ensure
that when other devices are making heavy use of your internet connection
VOIP packets don't get dropped.

[David Woolley]

Post by Brian Gregory
You just need your VOIP device to make an outgoing connection and keep
it alive.

Which is not possible for media, as media uses connectionless UDP, and
you cannot force the far end to reserve a port number for you. Also,
although some providers will allow you to use TCP for signalling, the
provider sets up the call for incoming requests, so you can't control that.

[David Sankey]

Post by David Woolley

Post by Brian Gregory
You just need your VOIP device to make an outgoing connection and keep
it alive.

Which is not possible for media, as media uses connectionless UDP, and
you cannot force the far end to reserve a port number for you.  Also,
although some providers will allow you to use TCP for signalling, the
provider sets up the call for incoming requests, so you can't control that.

I have used a number of VoIP providers with a selection of devices,
Cisco ATA and Gigaset, over ADSL, VDSL and FTTP, all with NAT on the
LAN, for 10 years now with no difficulty whatsoever.

It is one of those things that just works.

D

[Woody]

Post by David Sankey

Post by David Woolley

Post by Brian Gregory
You just need your VOIP device to make an outgoing connection and
keep it alive.

Which is not possible for media, as media uses connectionless UDP, and
you cannot force the far end to reserve a port number for you.  Also,
although some providers will allow you to use TCP for signalling, the
provider sets up the call for incoming requests, so you can't control that.

I have used a number of VoIP providers with a selection of devices,
Cisco ATA and Gigaset, over ADSL, VDSL and FTTP, all with NAT on the
LAN, for 10 years now with no difficulty whatsoever.
It is one of those things that just works.
D

+1

[Brian Gregory]

Post by David Woolley

Post by Brian Gregory
You just need your VOIP device to make an outgoing connection and keep
it alive.

Which is not possible for media, as media uses connectionless UDP, and
you cannot force the far end to reserve a port number for you.  Also,
although some providers will allow you to use TCP for signalling, the
provider sets up the call for incoming requests, so you can't control that.

Well it works for me no matter whether David Woolley says it's impossible.

Yes, it all uses UDP.

[Marco Moock]

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and
keep it alive.

Many people have problems with VoIP in that case.
I haven't investigated that as I don't care about the dinosaur internet
protocol anymore when using VoIP.

[Brian Gregory]

Post by Marco Moock

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and
keep it alive.

Many people have problems with VoIP in that case.
I haven't investigated that as I don't care about the dinosaur internet
protocol anymore when using VoIP.

Presumably you still want a firewall though?

Just make suere you choose a modern VoIP provider where it really
doesn't make any difference whether you're behind NAT (IPv4) or behind a
firewall (IPv6).

[Marco Moock]

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and
keep it alive.

Many people have problems with VoIP in that case.
I haven't investigated that as I don't care about the dinosaur
internet protocol anymore when using VoIP.

Presumably you still want a firewall though?

SPI firewalls for IPv6 exists. Exceptions make VoIP very easy through
it.

Post by Brian Gregory
Just make suere you choose a modern VoIP provider where it really
doesn't make any difference whether you're behind NAT (IPv4) or
behind a firewall (IPv6).

Try it out with IPv4 CGNAT and make the bad experience yourselves.

[Brian Gregory]

Post by Marco Moock

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and
keep it alive.

Many people have problems with VoIP in that case.
I haven't investigated that as I don't care about the dinosaur
internet protocol anymore when using VoIP.

Presumably you still want a firewall though?

SPI firewalls for IPv6 exists. Exceptions make VoIP very easy through
it.

Do you allow VoIP in from any IP or do you resolve the domain name of
your VoIP provider and only allow VoIP from that IP (or group of IPs)
in, hoping that it doesn't change to a different IP?

Post by Marco Moock

Post by Brian Gregory
Just make suere you choose a modern VoIP provider where it really
doesn't make any difference whether you're behind NAT (IPv4) or
behind a firewall (IPv6).

Try it out with IPv4 CGNAT and make the bad experience yourselves.

Why would CGNAT be any worse than the NAT in my router?

[Theo]

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and
keep it alive.

Many people have problems with VoIP in that case.
I haven't investigated that as I don't care about the dinosaur
internet protocol anymore when using VoIP.

Presumably you still want a firewall though?

SPI firewalls for IPv6 exists. Exceptions make VoIP very easy through
it.

Do you allow VoIP in from any IP or do you resolve the domain name of
your VoIP provider and only allow VoIP from that IP (or group of IPs)
in, hoping that it doesn't change to a different IP?

I've not tried this specifically but I wouldn't allow VOIP in, but the
stateful firewall means you can allow VOIP out (the registration) and then
track the connections coming in (the media) to allow them.

Allowing registration in from the internet is a recipe for spam. If your
VOIP provider makes inbound connections they'll provide you with a list of
IPs to whitelist in the firewall.

Most domestic scale VOIP is primarily outbound (your ATA makes the outbound
connections to the SIP server, and media goes the same way) so there's no
specific firewalling needed AIUI. With a STUN server set it works over NAT.

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
Just make suere you choose a modern VoIP provider where it really
doesn't make any difference whether you're behind NAT (IPv4) or
behind a firewall (IPv6).

Try it out with IPv4 CGNAT and make the bad experience yourselves.

Why would CGNAT be any worse than the NAT in my router?

CGNAT is one giant NAT box for thousands of your ISP's customers, so if it
breaks or goes slow, so does your internet. Often they are underpowered for
the number of connections using them. If your router misbehaves just reboot
it, but if the CGNAT misbehaves you have to call customer services and tell
them, yes you have reinstalled Windows but it's still doing it...

Also, the CGNAT means sharing an IP with thousands of other people. If one
of those gets the IP banned from a site, so is everyone else also banned.
Or if not banned then more 'we noticed you doing something suspicious'
CAPTCHAs and random hurdles like insisting you verify a phone number before
proceeding.

Theo

[Brian Gregory]

Post by Theo
I've not tried this specifically but I wouldn't allow VOIP in, but the
stateful firewall means you can allow VOIP out (the registration) and then
track the connections coming in (the media) to allow them.

Exactly.

You only allow it on from where you outgoing connection was going to,
and you have a timeout.

This is what I have been saying, just get your device to do keep alive
to keep the return path open.

Post by Theo
Allowing registration in from the internet is a recipe for spam. If your
VOIP provider makes inbound connections they'll provide you with a list of
IPs to whitelist in the firewall.

Makes sense, but I've never had to do that.

Post by Theo
Most domestic scale VOIP is primarily outbound (your ATA makes the outbound
connections to the SIP server, and media goes the same way) so there's no
specific firewalling needed AIUI. With a STUN server set it works over NAT.

So far I've always found STUN unnecessary.
In my router I just disable SIP ALG (Application Layer Gateway), enable
RTSP ALG, and that's all. Nothing else is needed.

In my VoIP device I just enable keep alive and set it to less than my
router's UDP connection timeout. Nothing else special is needed.

[Pancho]

Post by Theo
Allowing registration in from the internet is a recipe for spam. If your
VOIP provider makes inbound connections they'll provide you with a list of
IPs to whitelist in the firewall.

I left VoIP port 5060 open for about a decade, the only real problem I
had was phantom calls in the middle of the night.

So I did set up a whitelist, as you suggest.

Post by Theo
Most domestic scale VOIP is primarily outbound (your ATA makes the outbound
connections to the SIP server, and media goes the same way) so there's no
specific firewalling needed AIUI. With a STUN server set it works over NAT.

SIP isn't the problem. Media, RTP, is the problem. If what you say is
true, it requires a non-NAT intermediary server, i.e. calls cannot be
direct P2P, this increases latency.

The need for STUN tends to contradict your assertion that "your ATA
makes the outbound connection". The STUN problem is that the SIP
handshaking message sends the ATA's IP address, and invites the remote
end to set up an RTP connection to that ATA address. If the ATA is
behind NAT the address will be a LAN address (e.g. 192.168.0.x), which
is meaningless to the remote end. This suggests the remote end (Server
end) is opening a connection to the ATA.

Most modern routers have a SIP Application Layer Gateway (ALG), it
inspects packet headers for a SIP tag and modifies this SIP message to
send the external IP address, instead of the local one. It must do some
other stuff too, so the incoming connection request goes to the correct
LAN IP, (assuming the ports aren't already forwarded). These ALGs are
notoriously problematic. This is the real difference between NAT/IPv4
and IPv6 with IPv6 the internal IP address is meaningful externally.

SIP/RTP is just shit in the NAT world.

[Brian Gregory]

Post by Pancho

Allowing registration in from the internet is a recipe for spam.  If your
VOIP provider makes inbound connections they'll provide you with a list of
IPs to whitelist in the firewall.

I left VoIP port 5060 open for about a decade, the only real problem I
had was phantom calls in the middle of the night.
So I did set up a whitelist, as you suggest.

Most domestic scale VOIP is primarily outbound (your ATA makes the outbound
connections to the SIP server, and media goes the same way) so there's no
specific firewalling needed AIUI.  With a STUN server set it works
over NAT.

SIP isn't the problem. Media, RTP, is the problem. If what you say is
true, it requires a non-NAT intermediary server, i.e. calls cannot be
direct P2P, this increases latency.

They are always via an intermediary. That's how VoIP is done now.
If I send my SIP command to call someone to sipxxx.voipservice.net then
my RTP will also be via connections between me and say,
rtpnnn.voipservice.net.

Post by Pancho
The need for STUN tends to contradict your assertion that "your ATA
makes the outbound connection".

You don't need STUN nowadays.

Post by Pancho
Most modern routers have a SIP Application Layer Gateway (ALG), it
inspects packet headers for a SIP tag and modifies this SIP message to
send the external IP address, instead of the local one.

I have always found SIP ALG just messes everything up and needs to be
turned off.

Post by Pancho
It must do some
other stuff too, so the incoming connection request goes to the correct
LAN IP, (assuming the ports aren't already forwarded). These ALGs are
notoriously problematic. This is the real difference between NAT/IPv4
and IPv6 with IPv6 the internal IP address is meaningful externally.
SIP/RTP is just shit in the NAT world.

I've always found it trivially easy to set up and get working.

[Marco Moock]

Post by Brian Gregory
They are always via an intermediary. That's how VoIP is done now.
If I send my SIP command to call someone to sipxxx.voipservice.net
then my RTP will also be via connections between me and say,
rtpnnn.voipservice.net.

I've never seen that.
At least Linphone and AVM devices have 2 channels: SIP for signaling
and RTP for speech.
When using NAT, the ALG needs to investigate that and create the
matching rulesets.

[Brian Gregory]

Post by Marco Moock

Post by Brian Gregory
They are always via an intermediary. That's how VoIP is done now.
If I send my SIP command to call someone to sipxxx.voipservice.net
then my RTP will also be via connections between me and say,
rtpnnn.voipservice.net.

I've never seen that.
At least Linphone and AVM devices have 2 channels: SIP for signaling
and RTP for speech.
When using NAT, the ALG needs to investigate that and create the
matching rulesets.

You're saying your RTP packets go directly to / come directly from the
person you are calling?

[Marco Moock]

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
They are always via an intermediary. That's how VoIP is done now.
If I send my SIP command to call someone to sipxxx.voipservice.net
then my RTP will also be via connections between me and say,
rtpnnn.voipservice.net.

I've never seen that.
At least Linphone and AVM devices have 2 channels: SIP for signaling
and RTP for speech.
When using NAT, the ALG needs to investigate that and create the
matching rulesets.

You're saying your RTP packets go directly to / come directly from
the person you are calling?

No, they go to my VoIP provider.

[Pancho]

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
They are always via an intermediary. That's how VoIP is done now.
If I send my SIP command to call someone to sipxxx.voipservice.net
then my RTP will also be via connections between me and say,
rtpnnn.voipservice.net.

I've never seen that.
At least Linphone and AVM devices have 2 channels: SIP for signaling
and RTP for speech.
When using NAT, the ALG needs to investigate that and create the
matching rulesets.

You're saying your RTP packets go directly to / come directly from the
person you are calling?

I'm saying they should be able to. When VoIP was younger it offered the
promise of direct IP dialling, and third party directories, that enabled
P2P connections (in effect directories gave you the IP to dial). Phones
like Siemens Gigaset offered a service to do this Gigaset.net.

There were strong reasons why this was a good idea. Simple directory
services are clearly cheaper in hardware terms than relays for RTP
streams. P2P connections are more private, less prone to government
snooping. P2P connections have lower latency, less echo.

With modern, faster, connections latency/echo is less of a problem. Even
so I'm thinking of setting up Asterisk to enable P2P connections with
close contacts, i.e. I run an Asterisk server and they register on it.
Directmedia should allow even the Asterisk server to be cut out of the
RTP connections. To be fair, I have been thinking of doing this for
years, and not got around to it.

The rise of free mobile minutes, makes VoIP less interesting. So I'll
probably never bother. My main telecom wish now is to have multiple IP
phones so I can ditch DECT. But..., on the other hand, Android blocks
phone call recording, so maybe I will set up Asterisk to do this on VoIP.

[Rupert Moss-Eccardt]

Post by Pancho

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
They are always via an intermediary. That's how VoIP is done now.
If I send my SIP command to call someone to sipxxx.voipservice.net
then my RTP will also be via connections between me and say,
rtpnnn.voipservice.net.

I've never seen that.
At least Linphone and AVM devices have 2 channels: SIP for signaling
and RTP for speech.
When using NAT, the ALG needs to investigate that and create the
matching rulesets.

You're saying your RTP packets go directly to / come directly from the
person you are calling?

I'm saying they should be able to. When VoIP was younger it offered the
promise of direct IP dialling, and third party directories, that enabled
P2P connections (in effect directories gave you the IP to dial). Phones
like Siemens Gigaset offered a service to do this Gigaset.net.
There were strong reasons why this was a good idea. Simple directory
services are clearly cheaper in hardware terms than relays for RTP
streams. P2P connections are more private, less prone to government
snooping. P2P connections have lower latency, less echo.
With modern, faster, connections latency/echo is less of a problem. Even
so I'm thinking of setting up Asterisk to enable P2P connections with
close contacts, i.e. I run an Asterisk server and they register on it.
Directmedia should allow even the Asterisk server to be cut out of the
RTP connections. To be fair, I have been thinking of doing this for
years, and not got around to it.
The rise of free mobile minutes, makes VoIP less interesting. So I'll
probably never bother. My main telecom wish now is to have multiple IP
phones so I can ditch DECT. But..., on the other hand, Android blocks
phone call recording, so maybe I will set up Asterisk to do this on VoIP.

And "industrial" deployments do that - go to the target SBC without
passing through a core

[Marco Moock]

Post by Theo
I've not tried this specifically but I wouldn't allow VOIP in, but the
stateful firewall means you can allow VOIP out (the registration) and
then track the connections coming in (the media) to allow them.

This only works if SIP is unencrypted because the SPI FW needs to
examine the port for the RTP voice channel. And it only works if such
gateway is available, which is often not the case.

[Pancho]

Post by Marco Moock

Post by Theo
I've not tried this specifically but I wouldn't allow VOIP in, but the
stateful firewall means you can allow VOIP out (the registration) and
then track the connections coming in (the media) to allow them.

This only works if SIP is unencrypted because the SPI FW needs to
examine the port for the RTP voice channel. And it only works if such
gateway is available, which is often not the case.

Actually, I don't think that is what stateful means. I speak from my
usual lofty "jack of all trades" ignorance, so feel free to contradict
me, but I think stateful only works with one connection, one path, i.e.
if you connect to a remote IP/port from a local IP/port, the firewall
remembers to allow packets making the return trip, from the remote IP.

The problem with SIP/RTP is that the RTP connections work on different
ports to the initial SIP connection, they take different paths, so the
stateful firewall doesn't have anything to work with.

It is possible the VoIP ALG deals with this, interpreting the SIP
packets and using the information to open the firewall for the RTP
streams, but I don't think that is classified as stateful.

As I said there is a huge caveat that everything I know about VoIP is
from 15 years ago, and I never really understood it then. In the interim
I have just hoped it would die and be replaced by something better, but
like unencrypted email it just lingers around. The good, better,
technologies are all proprietary.

[Marco Moock]

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
I receive calls fine behind NAT and a firewall.
I don't believe CGNAT would be a problem either.
You just need your VOIP device to make an outgoing connection and
keep it alive.

Many people have problems with VoIP in that case.
I haven't investigated that as I don't care about the dinosaur
internet protocol anymore when using VoIP.

Presumably you still want a firewall though?

SPI firewalls for IPv6 exists. Exceptions make VoIP very easy
through it.

Do you allow VoIP in from any IP or do you resolve the domain name of
your VoIP provider and only allow VoIP from that IP (or group of IPs)
in, hoping that it doesn't change to a different IP?

I allow it from all IPs and have a static port for the incoming RTP.

Post by Brian Gregory

Post by Marco Moock

Post by Brian Gregory
Just make suere you choose a modern VoIP provider where it really
doesn't make any difference whether you're behind NAT (IPv4) or
behind a firewall (IPv6).

Try it out with IPv4 CGNAT and make the bad experience yourselves.

Why would CGNAT be any worse than the NAT in my router?

I dunno the exact reason (I hate those old-ass crap technology, so I
use IPv6), but my experience confirms that with CGNAT incoming VoIP
connections are a PITA. I use my VPN when being remote and needing to
phone somebody.

[Brian Gregory]

Post by Marco Moock
I allow it from all IPs and have a static port for the incoming RTP.

See Theo's answer above.

[Marco Moock]

Post by Brian Gregory

Post by Marco Moock
I allow it from all IPs and have a static port for the incoming RTP.

See Theo's answer above.

I've already tested that. It only works if the SIP connection is
unencrypted.
https://lists.fedorahosted.org/archives/list/firewalld-***@lists.fedorahosted.org/thread/CSEIKX6626T2RXNHVBAGK7OHZWYH24UR/

[David Woolley]

Post by Brian Gregory
Why would CGNAT be any worse than the NAT in my router?

Basically because CGNAT is a technology for consumers, but VoIP is
designed to be run on servers; consumers aren't expected to run servers.

The problem is that you don't have control over the port forwarding
rules, and because UDP is connectionless, unless the ISP specifically
recognizes SIP and looks inside it, you tend to get the default
assumptions for UDP which are that consumers don't receive unsolicited
messages expect prompt replies to outgoing messages, and don't expect
information sent to one port to affect transfers over another port, so
may translate the IP address differently, and may translate the port number.

The standard SIP session establishment involves the port numbers and IP
addresses to be used being sent using different a message with different
port numbers.

This means you have to use tricks like make work traffic to let the
system know that you are still using particular addresses, so the
network doesn't re-assign them to another customer, and the provider has
to wait for media, from you, to work out what the real port and IP
address are for media to you.

CGNAT is often used my mobile phone companies, whose only incentive to
look inside VoIP traffic is likely to be to identify it as such, so that
they can block it completely, rather than ensure it is protected from
gratuitous address changes.

The optimum NAT configuration for VoIP is that you set static port
forwarding rules for the signalling port, and for the range of media
ports in use, so that port numbers go through unchanged, and without
having to send traffic for the router to learn about them, but people
operating CGNAT do not give you any control over their routers.

If you are a home user with simple NAT, you can probably use static
rules, or can, at least, tweak the timeouts for automatic rules, so once
learned, they are effectively static. There is also unlikely to be a
collision with regards to the source port, so the router may well set up
its learned rules without actually translating the source ports. There
is typically only one IP address involved, so that devices can use
protocols to determine that, with an expectation that that address will
apply to all their external traffic until the provider next messes with
their IP address (which unfortunately many providers do - again not
expecting people to need addresses to be valid beyond a single web
download, or movie viewing).

If you are using FreePBX (or bare Asterisk) to run a home PABX, the
standard setup involves configuring it with the public addresses, and
assuming port numbers are untranslated. You can run it with unstable
addresses, and translated port numbers, but it is likely to break for
several minutes if the provider changes any of these, until it relearns
its own address and re-registers with the provider.

[David Woolley]

Post by Brian Gregory
Do you allow VoIP in from any IP or do you resolve the domain name of
your VoIP provider and only allow VoIP from that IP (or group of IPs)
in, hoping that it doesn't change to a different IP?

ITSPs normally have a restricted block of addresses from which they send
traffic, which shouldn't change, so a simple home user would be advised
to white list just that block. The block may not correspond to the any
of the published domain names, and domain name lookups for VoIP are
often not simple A record lookups, but SVR ones with subsequent lookups
to get the actual address.

Power home users and business users, often want to have people access a
PABX, either located on site, or in the cloud, from external locations.
The ideal solution there is VPNs (not the identity hiding products, but
the original meaning of VPN), but that is not always possible. In which
case they may try white listing the ISPs used by their users, black
listing known telephone toll fraud countries, and ISPs, or a combination
of these and software, like fail2ban, that trawls logs, and dynamically
adds firewall rules to block addresses that have generated suspicious
traffic, e.g. authentication failures. (Other tactics involve
registering from non-standard port numbers, or using less common protocols.)

Phones which aren't firewalled are likely to receive phantom calls, but
these calls are generally trying to find PABXes, for toll fraud
purposes, rather than do voice spam.

(Historically, SIP was designed to connect direct to the destination
organisation (only requiring TCP/IP level infrastructure), rather than
mimic the exchange hierarchy of the traditional PSTN, but almost nobody
uses that capability, as is also the case with email.)

[Richmond]

Post by Spike
A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.
- finding a suitable VOIP provider
- finding suitable equipment, such as an ATA or modem/router with it built
in (uncommon)
- setting up the equipment with the correct information
- ‘tweaking’ the equipment settings
- ‘tweaking’ the supplier’s settings
- much testing to ensure it actually works properly
- finding that in some cases some settings aren’t tweakable.
One forms the opinion that the exercise bears some similarity to banging
one’s head against a wall, always presuming you can find the right sort to
bang against.
In the presence of a well-developed mobile phone system, which for the vast
majority of people ‘just works’, why bother with VOIP? It seems like a
futile and nerdy exercise in clinging on to something that is well past its
time and which doesn’t translate well into the 21st century.
YMMV

An easy way to get started, if you have a smart phone, is to use
voipfone.co.uk and the voipfone app called Voipfone Softfone. It doesn't
require any configuration beyond a username and password. Wifi has extra
latency I think but it's ok.

Having done that I subsequently got a router with a phone socket and
configured that to use voipfone with a dect handset. It took some
experiments to make it work, mainly because the router is wierd, but now
I use it all the time for outgoing calls.

[Richmond]

Post by Richmond
voipfone.co.uk and the voipfone app called Voipfone Softfone. It doesn't

That should be Voipfone Softphone.

[David Wade]

Post by Spike
A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.
- finding a suitable VOIP provider

Not really. Plenty of options. ZEN, A&A , Voipfone.co.uk or BT/EE

Post by Spike
- finding suitable equipment, such as an ATA or modem/router with it built
in (uncommon)

I think these will become more common. The common 3rd party routera,
e.g. TPlink, Fritz!Box and Draytek have options with built-in VOIP

Post by Spike
- setting up the equipment with the correct information
- ‘tweaking’ the equipment settings
- ‘tweaking’ the supplier’s settings
- much testing to ensure it actually works properly
- finding that in some cases some settings aren’t tweakable.

Basically set up mine in 10 minutes.

Post by Spike
One forms the opinion that the exercise bears some similarity to banging
one’s head against a wall, always presuming you can find the right sort to
bang against.

Done no banging....

Post by Spike
In the presence of a well-developed mobile phone system, which for the vast
majority of people ‘just works’, why bother with VOIP? It seems like a
futile and nerdy exercise in clinging on to something that is well past its
time and which doesn’t translate well into the 21st century.

lots of people have my number. Modern 4g and 5G are really just VOIP
under the covers..

Post by Spike
YMMV

Dave

[Theo]

Post by Spike
A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.
- finding a suitable VOIP provider
- finding suitable equipment, such as an ATA or modem/router with it built
in (uncommon)
- setting up the equipment with the correct information
- ‘tweaking’ the equipment settings
- ‘tweaking’ the supplier’s settings
- much testing to ensure it actually works properly
- finding that in some cases some settings aren’t tweakable.

I've been using VOIP for almost 20 years and these days find it 'just
works'. Even using shady cheapskate providers for international call
routing.

(that said, I'm not starting from scratch)

Post by Spike
One forms the opinion that the exercise bears some similarity to banging
one’s head against a wall, always presuming you can find the right sort to
bang against.
In the presence of a well-developed mobile phone system, which for the vast
majority of people ‘just works’, why bother with VOIP? It seems like a
futile and nerdy exercise in clinging on to something that is well past its
time and which doesn’t translate well into the 21st century.

I think you could phrase it the other way. VOIP can give you a full
business telecoms platform - multiple numbers, multiple extensions,
lowest-cost call routing, voicemail to email, call recording, itemised
call logs, a wide variety of handsets (wired and portable)...

A mobile gives you a way to make phone calls. That's it really. The
handsets aren't really designed for making voice calls, call quality can be
variable, the carriers like to make extras chargeable, and when you go
outside of your 'plan' (eg international calls) it gets very expensive very
quickly.

Maybe those limits are fine for you, but I think may people may find some
VOIP features useful. eg having your voicemails arrive as emails rather
than having to dial in to your network's answering machine, or being able to
use a handset that's designed for voice calls rather than holding a sharp
piece of glass to your head.

I'm sure a lot of people aren't bothered because don't make many phone
calls so it doesn't really matter to them, and that's fair enough. It's
just nice to have options.

Theo

[Woody]

Post by Theo

Post by Spike
A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.
- finding a suitable VOIP provider
- finding suitable equipment, such as an ATA or modem/router with it built
in (uncommon)
- setting up the equipment with the correct information
- ‘tweaking’ the equipment settings
- ‘tweaking’ the supplier’s settings
- much testing to ensure it actually works properly
- finding that in some cases some settings aren’t tweakable.

I've been using VOIP for almost 20 years and these days find it 'just
works'. Even using shady cheapskate providers for international call
routing.
(that said, I'm not starting from scratch)

Post by Spike
One forms the opinion that the exercise bears some similarity to banging
one’s head against a wall, always presuming you can find the right sort to
bang against.
In the presence of a well-developed mobile phone system, which for the vast
majority of people ‘just works’, why bother with VOIP? It seems like a
futile and nerdy exercise in clinging on to something that is well past its
time and which doesn’t translate well into the 21st century.

I think you could phrase it the other way. VOIP can give you a full
business telecoms platform - multiple numbers, multiple extensions,
lowest-cost call routing, voicemail to email, call recording, itemised
call logs, a wide variety of handsets (wired and portable)...
A mobile gives you a way to make phone calls. That's it really. The
handsets aren't really designed for making voice calls, call quality can be
variable, the carriers like to make extras chargeable, and when you go
outside of your 'plan' (eg international calls) it gets very expensive very
quickly.
Maybe those limits are fine for you, but I think may people may find some
VOIP features useful. eg having your voicemails arrive as emails rather
than having to dial in to your network's answering machine, or being able to
use a handset that's designed for voice calls rather than holding a sharp
piece of glass to your head.
I'm sure a lot of people aren't bothered because don't make many phone
calls so it doesn't really matter to them, and that's fair enough. It's
just nice to have options.
Theo

Totally agree Theo, plus one extra. If you have a VoIP account you can
run an app on your mobile phone which allows you to make (and receive)
calls through your VoIP provider and thus get the benefits of the
cheaper call costs (especially international).

Another thing that there seems to be increasing use of is multiple
access. A chap I know has a shop locally but he lives in another small
town about 8 miles away. His business phone number is his home phone
number (different exchange) and he has an app on his mobile. Ergo during
the day anyone calling his home number will be answered on his mobile,
but whilst he is travelling to and from home his wife can answer calls
at home. It also means that he is not stuck to exact shop opening times
as his call is always answer by himself or herself.

[Woody]

Post by Woody

Post by Theo

Post by Spike
A reading of the posts in the group regarding setting up a VOIP system
seems to suggest the exercise has a number of pitfalls.
- finding a suitable VOIP provider
- finding suitable equipment, such as an ATA or modem/router with it built
in (uncommon)
- setting up the equipment with the correct information
- ‘tweaking’ the equipment settings
- ‘tweaking’ the supplier’s settings
- much testing to ensure it actually works properly
- finding that in some cases some settings aren’t tweakable.

I've been using VOIP for almost 20 years and these days find it 'just
works'.  Even using shady cheapskate providers for international call
routing.
(that said, I'm not starting from scratch)

Post by Spike
One forms the opinion that the exercise bears some similarity to banging
one’s head against a wall, always presuming you can find the right sort to
bang against.
In the presence of a well-developed mobile phone system, which for the vast
majority of people ‘just works’, why bother with VOIP? It seems like a
futile and nerdy exercise in clinging on to something that is well past its
time and which doesn’t translate well into the 21st century.

I think you could phrase it the other way.  VOIP can give you a full
business telecoms platform - multiple numbers, multiple extensions,
lowest-cost call routing, voicemail to email, call recording, itemised
call logs, a wide variety of handsets (wired and portable)...
A mobile gives you a way to make phone calls.  That's it really.  The
handsets aren't really designed for making voice calls, call quality can be
variable, the carriers like to make extras chargeable, and when you go
outside of your 'plan' (eg international calls) it gets very expensive very
quickly.
Maybe those limits are fine for you, but I think may people may find some
VOIP features useful.  eg having your voicemails arrive as emails rather
than having to dial in to your network's answering machine, or being able to
use a handset that's designed for voice calls rather than holding a sharp
piece of glass to your head.
I'm sure a lot of people aren't bothered because don't make many phone
calls so it doesn't really matter to them, and that's fair enough.  It's
just nice to have options.
Theo

Totally agree Theo, plus one extra. If you have a VoIP account you can
run an app on your mobile phone which allows you to make (and receive)
calls through your VoIP provider and thus get the benefits of the
cheaper call costs (especially international).
Another thing that there seems to be increasing use of is multiple
access. A chap I know has a shop locally but he lives in another small
town about 8 miles away. His business phone number is his home phone
number (different exchange) and he has an app on his mobile. Ergo during
the day anyone calling his home number will be answered on his mobile,
but whilst he is travelling to and from home his wife can answer calls
at home. It also means that he is not stuck to exact shop opening times
as his call is always answer by himself or herself.

I should have said that his home VoIP (don't know if it is a voip phone
or an ATA) and the app on his mobile have the same config so that both
will ring and either can answer an incoming call.

[www.GymRatZ.co.uk]

Post by Woody
Totally agree Theo, plus one extra. If you have a VoIP account you can
run an app on your mobile phone which allows you to make (and receive)
calls through your VoIP provider and thus get the benefits of the
cheaper call costs (especially international).
Another thing that there seems to be increasing use of is multiple
access. A chap I know has a shop locally but he lives in another small
town about 8 miles away. His business phone number is his home phone
number (different exchange) and he has an app on his mobile. Ergo during
the day anyone calling his home number will be answered on his mobile,
but whilst he is travelling to and from home his wife can answer calls
at home. It also means that he is not stuck to exact shop opening times
as his call is always answer by himself or herself.

Sounds like my scenario.
:)

For about the same cost as 2 accounts with 2 call barring features etc I
was able to set up 3 extensions on the main account which were all
protected by the single account call barring feature, and have the home
telephone number only routed to the home extension (after getting it
re-allocated from the original home voipfone account to the shop
account) so no-one calling home (parents primarily) noticed a
difference and only the home phones ring.
The main shop number is routed to all 3 extensions, i.e. home, shop, and
mobile (running voipfone soft-fone app) so even if away from any
property I can take calls to the shop number as long as I have wifi or
4G signal, and obviously at home I get shop calls but can also put "open
hours" on the shop number so we don't get business/pest calls past 21:00
or whatever although the call-barring is now so deeply trained after so
many years virtually no junk calls get through anyway.

I can also phone out on mobile through VoipFone app and present shop
number for caller I.D. more trustworthy to customers than a random
mobile number.

It's vastly better now than when I played with voipfone extensions about
15 years ago and it didn't serve any purpose or benefit at that time,
just more complications and cost.

[Mark Carver]

My view, for the great unwashed, probably not.

23 years ago, after an awful lot of rejections, because our phone line
was 5.8km long, I finally managed to get an ADSL connection, (ISP was
Pipex, remember them ?)

I bought this ADSL Router from Solwise:-

<https://www.solwise.co.uk/downloads/files/setup_for_sar715_on_8_2-vr5.pdf>

It was actually quite daunting to set all the correct parameters up, and
I was shocked that the connection burst into life on my first attempt.

(It had a default IP address of 192.168.7.1, and to this day, many
routers later, I still use that address on my home LAN)

Of course ADSL/VDSL took off, became mass market, and the set up of even
third party routers is so much easier now.

VoIP set up reminds me of those dark early 00s DSL days, the difference
is, it's not ever going to become mass market, and therefore, I wouldn't
expect set up of devices to get streamlined in the same way *DSL etc
routers have.

The so called 'Digital Voice' services embedded into ISP supplied
routers, are in my opinion the way to go for 95% of users (yes, I know
they will be paying through the nose for call charges etc), but I
suspect by 2034 virtually no one will be using 'landline' voice services
in a domestic setting.

[David Wade]

Post by Mark Carver
My view, for the great unwashed, probably not.
23 years ago, after an awful lot of rejections, because our phone line
was 5.8km long, I finally managed to get an ADSL connection, (ISP was
Pipex, remember them ?)
I bought this ADSL Router from Solwise:-
<https://www.solwise.co.uk/downloads/files/setup_for_sar715_on_8_2-vr5.pdf>
It was actually quite daunting to set all the correct parameters up, and
I was shocked that the connection burst into life on my first attempt.
(It had a default IP address of 192.168.7.1, and to this day, many
routers later, I still use that address on my home LAN)
Of course ADSL/VDSL took off, became mass market, and the set up of even
third party routers is so much easier now.
VoIP set up reminds me of those dark early 00s DSL days, the difference
is, it's not ever going to become mass market, and therefore, I wouldn't
expect set up of devices to get streamlined in the same way *DSL etc
routers have.

No reason why it can't be done...

Post by Mark Carver
The so called 'Digital Voice' services embedded into ISP supplied
routers, are in my opinion the way to go for 95% of users (yes, I know
they will be paying through the nose for call charges etc), but I
suspect by 2034 virtually no one will be using 'landline' voice services
in a domestic setting.

ZEN is charging around £5/month and you can just plug a phone into the
socket on their router.

Dave

[Theo]

Post by Mark Carver
VoIP set up reminds me of those dark early 00s DSL days, the difference
is, it's not ever going to become mass market, and therefore, I wouldn't
expect set up of devices to get streamlined in the same way *DSL etc
routers have.

VOIP *is* mass market. That market is businesses. Those of us doing VOIP
at home are just running a business-grade service at home, like some people
have pro grade camera or audio gear or whatever.

Businesses don't need 'VOIP for dummies' handholding, they pay people with
skills to install and configure it. Those skills are not taught only at
VOIP-Hogwarts, they aren't complicated and can be picked up by anyone
interested.

BTW, for consumer purposes the devices were streamlined 20 years ago by the
likes of Vonage: sign up for service, box arrives in the post, plug box
into router, plug phone into box, get phone calls. They still do that if
that's a service you want.

Post by Mark Carver
The so called 'Digital Voice' services embedded into ISP supplied
routers, are in my opinion the way to go for 95% of users (yes, I know
they will be paying through the nose for call charges etc), but I
suspect by 2034 virtually no one will be using 'landline' voice services
in a domestic setting.

As usual it depends what you need. For many people their ISP 'landline'
and/or their mobile setup is sufficient. If you have a good reason why
that one size doesn't fit your needs (reasons already mentioned in this
thread), then VOIP is another option.

By 2034 I think there will be increasing crossover between domestic and
business telecoms, for those with a business connection (eg run a small
business, work from home, freelancers, etc). It remains to be seen whether
that continues to be 'phone calls' or more takeup of
Teams/Zoom/Meet/Slack/... style communication platforms, but I think it is
unlikely that's going to be purely mobiles aside from one-man-bands.

Even then, the people they are calling will likely be using VOIP somewhere
in the mix - the company telecoms will run on VOIP whether or not they
have an extension at home, or their VOIP extension will actually ring their
Teams, or whatever.

TL;DR probably the majority of phone calls these days have a business at one
end or the other. VOIP still has plenty of relevance to that market,
especially for people who work for those businesses.

Theo

[noel]

Post by Theo

Post by Mark Carver
VoIP set up reminds me of those dark early 00s DSL days, the difference
is, it's not ever going to become mass market, and therefore, I
wouldn't expect set up of devices to get streamlined in the same way
*DSL etc routers have.

VOIP *is* mass market. That market is businesses. Those of us doing
VOIP at home are just running a business-grade service at home, like
some people have pro grade camera or audio gear or whatever.

-nod-
VoIP is also the only method being used for telco services in some
countries, USA, Australia (although the old POTS exchanges are remaining
active for time being in very rural areas in AU, all metro areas had the
last copper exchanges turned off about 2 or so years ago.

Don't think the UK wont go that way in the future, there is every
possibility it will, and yes I do wish they kept the copper exchanges
and the VDSL fibre to node/cabinet just passes on the X pairs back to it,
like in parts of Europe I understand, but, govt penny pinchers made sure
that didnt happen, so if your internet goes bung, tuff shit, find a
mobile and run up the hill to try get cell service /sigh/

Post by Theo
Businesses don't need 'VOIP for dummies' handholding, they pay people
with skills to install and configure it. Those skills are not taught
only at VOIP-Hogwarts, they aren't complicated and can be picked up by
anyone interested.

VoIP is also substantially cheaper - no line rentals, much cheaper calls,
most providers give free calls within their own network.

Advantage for businesses is you can have multiple providers and set up
least cost routing for prefixes you can get cheaper on one deal than the
other that might offer better pricing for other prefixes.

[David Woolley]

Post by noel
Don't think the UK wont go that way in the future, there is every
possibility it will

Currently scheduled for Autumn 2025. We've already passed the stop sell
date for analogue and ISDN phone services, so you will be forced to VoIP
if you move a service, it breaks, or you order a new service.

[noel]

Post by David Woolley

Post by noel
Don't think the UK wont go that way in the future, there is every
possibility it will

Currently scheduled for Autumn 2025. We've already passed the stop sell
date for analogue and ISDN phone services, so you will be forced to VoIP
if you move a service, it breaks, or you order a new service.

Thank's David, I'll add the UK to my list now :)

Cheers

[Mike Humphrey]

Post by David Woolley
Currently scheduled for Autumn 2025. We've already passed the stop sell
date for analogue and ISDN phone services, so you will be forced to VoIP
if you move a service, it breaks, or you order a new service.

In most, but not quite all, cases. I got a new analogue line installed
well after the stop-sell, as Openreach couldn't offer any other option.
And it sounds like there's still going to be cases where an analogue line
is the only option even after 2025 - just going back to a VoIP adaptor in
the exchange rather than traditional exchange kit.

What's changed is that you can't *choose* to have an analogue line if
there's a FTTP/FTTC option available. The same goes for ADSL - it's still
being sold, but only where no FTTC is available.

Mike

[David Wade]

Post by Mike Humphrey

Post by David Woolley
Currently scheduled for Autumn 2025. We've already passed the stop sell
date for analogue and ISDN phone services, so you will be forced to VoIP
if you move a service, it breaks, or you order a new service.

In most, but not quite all, cases. I got a new analogue line installed
well after the stop-sell, as Openreach couldn't offer any other option.
And it sounds like there's still going to be cases where an analogue line
is the only option even after 2025 - just going back to a VoIP adaptor in
the exchange rather than traditional exchange kit.

If that was before 1st December that was the only solution BT could
offer. The SOTAP product which delivers VOIP over ADSL didn't go live
until the 1st December..

https://www.ispreview.co.uk/index.php/2023/11/openreach-launch-all-ip-solution-for-uk-adsl-broadband-lines.html

Post by Mike Humphrey
What's changed is that you can't *choose* to have an analogue line if
there's a FTTP/FTTC option available. The same goes for ADSL - it's still
being sold, but only where no FTTC is available.

I think since 1st December no ADSL and no unbundling in the Exchange

Post by Mike Humphrey
Mike

Dave

[Mike Humphrey]

Post by David Wade
If that was before 1st December that was the only solution BT could
offer. The SOTAP product which delivers VOIP over ADSL didn't go live
until the 1st December..
https://www.ispreview.co.uk/index.php/2023/11/openreach-launch-all-ip-

solution-for-uk-adsl-broadband-lines.html

It was before 1st December. But SOTAP wouldn't have worked as there's no
usable ADSL either. Openreach are now installing fibre, so that site is
dealt with. But I still have one on my list - ADSL speed claimed as 0.1M
max, no FTTx. Ethernet theoretically available at an install cost of about
£500k. I strongly suspect we're going to end up with "SOTAP for
Analogue" (i.e. stick an ATA in the exchange building), but Openreach seem
reluctant to admit this.

Post by David Wade

Post by Mike Humphrey
What's changed is that you can't *choose* to have an analogue line if
there's a FTTP/FTTC option available. The same goes for ADSL - it's
still being sold, but only where no FTTC is available.

I think since 1st December no ADSL and no unbundling in the Exchange

Last time I looked it was March 2024. And what a surprise, it's been
extended again to March 2025!

https://www.ispreview.co.uk/index.php/2024/02/openreach-extend-uk-
analogue-line-rental-stop-sell-exemptions-by-1-year.html


Mike

[T i m]

On 29/03/2024 21:52, Theo wrote:

<snip>

Post by Theo
BTW, for consumer purposes the devices were streamlined 20 years ago by the
likes of Vonage: sign up for service, box arrives in the post, plug box
into router, plug phone into box, get phone calls. They still do that if
that's a service you want.

<snip>

That service was offered when I signed up to A&A recently. If you bought
the hardware from them, they would pre-configure it for you.

I had two Sipgate lines on my Fritz!Box FON WAN router for *years* and
whist it seemed daunting to configure back then, compared with the Cisco
and Yealink units, it was very simple!

Cheers, T i m

[Marco Moock]

Post by Mark Carver
VoIP set up reminds me of those dark early 00s DSL days, the
difference is, it's not ever going to become mass market, and
therefore, I wouldn't expect set up of devices to get streamlined in
the same way *DSL etc routers have.

In Germany, VoIP is mass-market and consumer internet access devices
support it out of the box with an easy setup.

[Theo]

Post by Mark Carver
The so called 'Digital Voice' services embedded into ISP supplied
routers, are in my opinion the way to go for 95% of users (yes, I know
they will be paying through the nose for call charges etc), but I
suspect by 2034 virtually no one will be using 'landline' voice services
in a domestic setting.

A belated comment as to the 'just use your mobile' idea. My mobile works at
home, and claims to have good signal. Texts work. But calls break up
randomly, perhaps because the signal isn't as good as good as it claims.
Result is people call me, I answer, we talk, but we end up saying 'sorry I
didn't catch that, can you repeat?' all the time. This is worse than having
no signal at all, when at least they'd try another way.

Meanwhile VOIP is crystal clear and Just Works.

(My mobile operator seemingly can't be bothered to implement wifi calling on
my tariff, which might or might not help signal problems)

Theo