Building a Personal VoIP System

Ah good memories! Even though I knew nothing about VoIP I installed, years ago, one at my wife&#x27;s little SME using &quot;RasPBX&quot; (a distro made of Raspbian + FreePBX + Asterisk) running on a... Raspberry Pi 1 (maybe a Pi 2 but I&#x27;m pretty sure it was a 1). The Pi was booting from the SD card but everything was running on an external HDD.<p>I did put six Cisco VoIP phones and all was working fine as long as no more than four phones were used simultaneously (which never happened). It worked for years like that and wife ended up selling her SME with these VoIP phones still hooked to the RPi 1. We warned them that that thing was kinda a hack ; )<p>For anyone hesitating: it&#x27;s not hard to set up. Find a provider to get a SIP trunk, configure the thing, backup the config (I just imaged the entire drive), and you&#x27;re good to go for a very long time.<p>It&#x27;s simple, reliable, stuff that usually won&#x27;t move under your feet.

I put in an asterix system 20 years ago, it still runs great, all I&#x27;ve done has been to replace a couple of dying (RAIDed) drives. I live in NZ used to live in the Bay Area, NZ trunks come in from a local SIP provider and use an Ooma box for our old US phone number, in home we had 5 FXSs and a sip phone in my office - everyone had a phone in their bedroom, and a couple on public spaces.<p>It&#x27;s set up so that all incoming calls hit a voice prompt saying which 1-digit extension you should enter to get who (this stops 99% of phone spam) - everyone gets a voicemail on their extension which rings in their bedroom and rolls over to the common spaces - and everyone gets a unique ring cadence, my daughter who at one point got 80% of the phone calls got the 2 short rings.<p>The kids have gone, we&#x27;ve moved to a smaller house, only 2 extensions, but that same hardware soldiers on (and still stops 99% of the spam - I&#x27;ve had 1 call in the past year)

Cool to read someone indulge in this. For others interested with no prior context, also consider FreeSWITCH.<p>Personal experience:<p>I spent about 15 months working for a local telecom company, supporting their asterisk servers and developing interactive voice response applications. I was given the opportunity to build their residential voip service (in 2010?) from scratch and despite asterisk being so dominant I prototyped and eventually completed the project using freeSWITCH. I found it to be so much more developer friendly to configure and extend. Being able to build dial plans and implement logic with JavaScript or Lua rather than asterisk&#x27;s config files was worth it. I suspect this system is still running.

&gt; Use a SIP Application Layer Gateway. This is a horrible feature offered by some routers. Basically, it deep-packet-inspects your SIP traffic, rewrites the headers, and creates port forwards on-the-fly to make sure the inbound audio stream makes its way to your device. SIP ALGs are a total hack and notoriously buggy.<p>Yes, these hacky ALG features also allowed internet users to access internal IPs on arbitrary services (!); named “remote arbitrary firewall pinhole control”. The attack was published in 2020 and named NAT slipstreaming [1].<p>[1] <a href="https:&#x2F;&#x2F;samy.pl&#x2F;slipstream&#x2F;" rel="nofollow">https:&#x2F;&#x2F;samy.pl&#x2F;slipstream&#x2F;</a>

Having gone down this road many times with freepbx, asterisk, etc.. I ultimately settled on just using voip.ms and connecting phones&#x2F;sip clients directly to their internal sub-accounts with voicemail. They have enough features for most users so you don&#x27;t need to worry about running your own PBX.

It&#x27;s worth noting that Asterisk will run on very low-end hardware: for the last 7 or so years, I&#x27;ve been running a small Asterisk box on a VIA C3 Mini-ITX system. System idle power is around 10W. The only reason I chose the VIA C3 system over something newer was that a) I already had it and b) it had a PCI slot (specifically PCI, not PCIe).<p>The PCI slot let me plug in a Digium TDM800P and add eight POTS lines, either FSX or FXO, for pretty cheap.

I’ve used to live in an inter generational big house (Italians am I right… even if we are not actually born there we still have spaghetti dna). Anyway I’ve setup one very lightweight asterix instance of statically defined accounts. A few cheap voip phones were installed through the house plus soft phones on personal smartphones.<p>It was supposed to make things more convenient. Supposed to, because we just kept yelling to convey messages. When I’ve moved out I’ve dismantled the system since I wouldn’t be there to keep it maintained.

&gt; SIP was initially released in 1999, and was designed with the assumption that each device has its own globally routable public IP address. After all, the IPv6 standard was released back in 1995, and NAT would soon be a thing of the past…right? Unforunately, this did not end up being the case.<p>AFAIK, most residential <i>and</i> commercial ISPs these days do assign customers both a dynamically-DHCP-leased IPv4 <i>address</i>, and a <i>static</i>, globally-routable IPv6 <i>prefix</i> — usually a &#x2F;64, though some are nicer than that. If you put your ISP&#x27;s gateway router into bridge mode, and then plugged your computer directly into it — then your device would acquire both an IPv4 and an IPv6 address.<p>But routers — including ISP gateway routers — insist on doing NAT not only for IPv4, but also for IPv6 (using the fe80:: prefix.) So on any regular home or office network, devices are going to acquire private-use IPv4 <i>and</i> IPv6 addresses.<p>Is there some reason that modern routers don&#x27;t do NAT for IPv4, while just further splitting+assigning the received prefix for IPv6, such that every device on the network receives a private IPv4 addr, but a <i>public</i> IPv6 prefix, e.g. a &#x2F;72?<p>I know that Internet-backbone network switches ignore the last 64 bits of IPv6 in their routing tables; but those bits are still being <i>carried</i> in the IPv6 packets, and once they reach your home router, <i>it</i> can make use of them to route to the final destination (i.e. one of the devices behind it.) Wasn&#x27;t this supposed to be the idea?

I&#x27;ve built my own business VoIP system — a few times over. I&#x27;ve used Asterisk and FreePBX (the free-ish GUI built on Asterisk), but settled on Asterisk for stability. At one point FreePBX pushed out an update that took down my system for a couple of days and baffled me until I read the FreePBX forums and saw similar complaints from other users.<p>If you want a week-ish long project, go for the full mid-2000s, boingboing and slashdot experience: buy the dead tree version of the O&#x27;Reilly Asterisk book, which has been kept up to date and is still an wonderful resource. Follow it until you have the Enterprise-grade phone system of your dreams.<p>I even hired Allison &quot;The Voice of Asterisk&quot; [0] to do our phone tree voice prompts. Her prices were very reasonable for a small batch of prompts, and it’s surreal hearing the same lady that does the IVR for your bank and cable company do your own voice prompts.<p>I use Twilio for my SIP trunking, and it has nice fallback features in case the Asterisk system needs to go down for maintenance or the like. Costs about $20&#x2F;month for a dozen or so users and fairly frequent calls&#x2F;SMS.<p>[0]: <a href="https:&#x2F;&#x2F;www.theivrvoice.com&#x2F;" rel="nofollow">https:&#x2F;&#x2F;www.theivrvoice.com&#x2F;</a>

This was sort of a thing in the mid-2000s, and I’m actually surprised to see it again. At the time I expected home routers to become SIP endpoints (and that came to pass with fiber), but they all expose FXO interfaces (jacks) rather than act as proxies for soft phones, so there’s a missed opportunity there. But the truth of the matter is that just shipping a DECT phone in a bundle is much easier for the carrier to troubleshoot.

I have been tinkering with a personal VoIP system in my spare time over the last couple of months. At this point, I have rescued the 3 lines of house wiring in my 1970s house and connected them to Asterisk on a VM via a Cisco MC3810 and a Adit 600 channel bank. This has involved messing with T1s which was a childhood dream of mine. I have 12 phones connected to the system, using my house wiring, a few point-to-point wires, and some SIP VoIP phones over ethernet. In turn these connect to Phreaknet, C*NET, and of course the normal PSTN via a pair of different ITSPs. I now feel almost qualified to build an early 2000s business voice phone system, for what very little that is worth. More importantly, I have a rotary phone on my desk. ;)

So, maybe one of the VoIP experts that showed up on this thread can help me with one doubt...<p>Let&#x27;s say that I have a LDAP server where I manage user accounts, and I want people to be able to call each other with any SIP-enabled phone. I am not interested in voicemail&#x2F;IVRs&#x2F;any type of &quot;voice application&quot; on top of that. Do I really need Asterisk&#x2F;FreeSWITCH or can I just go buy by setting up something like Kamailio?

my wife had a requirement: be able to have an intercom for the house and shed. We used to have a baby monitor, but that was one way, for one room. Now we need many more rooms to talk to each other.<p>I looked at some intercoms on amazon&#x2F;ebay, they are all RF and a bit shit. I saw some wifi ones, but nothing cheap enough to take a punt on. I did think about trying to make something with an ESP32, but that would be too hard for me in the time.<p>So I bought 6 cisco 7962 sip phones for £35 in total, and installed freepbx.<p>It took a bit of effort to bring the tftp server online, and make sure all the dhcp info was being passed on correctly. Once that was complete, freepbx makes most things pretty simple.<p>Now, I have ethernet is most rooms, and a switch that can do PoE, so this solution is for a niche of a niche

If only there was an equivalent for building your own home GSM&#x2F;LTE network. When around the house, I would love to connect to my own private cellular network. Not only is coverage poor where I live, but this would allow me to route calls in sophisticated and useful ways, because the backplane of modern cellular networks is VoIP.

The solution to the NAT issues is simply to use IPv6 and not worry about NAT.

Excellent article, and sections &quot;NAT Problems&quot; and &quot;NAT Solutions&quot; are a good starter on that topic.<p>Except even third-choice solution is not always feasible. Reserving fixed RTP&#x2F;UDP port range is not possible with carrier-grade NAT, which is quite common with residential ISPs and nearly-universal with cell ISPs.<p>Fourth-choice would be to reserve port range on a personal server (which would run B2BUA, asterisk in OP&#x27;s case; or an RTP proxy), and force calls, including media, from&#x2F;to SIP handsets to go via that.

There is SIP and XMPP, but they may be overconvoluted for what I am trying to achieve. Namely, I am not aiming at internet universality (no IPv4 sharing abomination built into the protocols).<p>I am looking at a modular set of protocols built mainly for IPv6.<p>The base: the &quot;telephone number&quot; would be ipv6:port. &quot;Ringing&quot; and video&#x2F;audio streams setup would be done here. End to end encrytion right from the start (only manual key exchange, zero automatic, even before ringing).<p>On top, a &quot;comfort protocol&quot;(one level of indirection) for those changing ipv6, but not &quot;accutely roaming&quot;, namely changing ipv6 while in a video&#x2F;audio call: a &quot;DynDNS&quot; but simpler, more a &quot;current IPv6:port of &#x27;name&#x27; kind of thing&quot;, &quot;address book with a drop of dynamic&quot;, &quot;name@server&quot; and you get the current ipv6:port. Unfortunately, it means &quot;accounts&quot; and real time updates. Of course, &quot;server&quot; could be a local&#x2F;dns&#x2F;ipv6. I am thinking zero password, only a public key.<p>For video&#x2F;audio streams, I may not bother and go TCP. The main constraint would be the timing information shared among video&#x2F;audio streams. No &quot;internet weather&quot; dynamic reconfiguration.<p>I even consider going &quot;horribly horrible&quot; for internet: idiotic binary based instead of text based protocols.

I have a VoIP system at home as well. I first used an RPi with Asterisk, but later switched to a Yeastar box with FXS ports (to connect a couple of Old School wired phones).<p>One thing that is making me REALLY MAD is that there are NO IPV6 TRUNK PROVIDERS in the US. Not a single one. At least none where I can just enter my credit card and get a phone line.<p>Somehow, the protocol designed to restore the end-to-end connectivity is not used for the poster child of end-to-end connectivity.

You can install Acrobits Groundwire or Bria. Those support PUSH notification for incoming calls. Push is better than missing calls because the app got killed, or forcing the app to run 24&#x2F;7 and severely shortening battery runtime. But the call quality will never be as good as the native phone app as that gets QCI prioritization.

I am one of the few people from my generation to maintain a &quot;land line&quot; (VoIP) and I, too, run it with Asterisk and the FreePBX configuration GUI. FreePBX provides a ton of macros out-of-the-box so that adding unwanted callers to a blacklist or doing phone number lookups is simple, for example. Why bother? Well, I find it interesting and fun, but most of the stuff I like can also be done with Google Voice. So I don&#x27;t recommend self-hosting a phone system unless you are really into the idea and want to spend a decent amount of time learning telecom domain knowledge.

Has someone good experience with a VoIP-Provider (not OVH or Sipgate) in Europe?

VoIP used to be standard on phones, even mobile smartphones. So I setup Asterisk so that family members could call each others as long as they where on WIfi. Unfortunately VoIP is no longer a built in standard so you need to download an app to use it. Before you could just dail 1 and my phone would ring, dail 2 and you would reach my wife.

I&#x27;d rather stab myself with WebRTC clients &amp; a signalling+TURN server instead of enduring Asterisk

Would be interesting to hear why they use v4+nat and face the described problem. First thought was maybe they bought some very old phones without v6 support - but the models he recommends don&#x27;t seem to have this problem based on a googled datasheet.

Great explanation of the SIP protocol. I’ve been looking for something that explains it this clearly forever, and this gets right to the point with the high level concept instead of getting immediately bogged down in technical specs.

Any recomendable sip Clients for iPhones to Connect to such a system?

People still actually make phone calls? I mean its neat that you can run your own voip, but I struggle to imagine it getting used much or being worth even $5&#x2F;month

Nice article. Is there a benefit to using a queue for incoming calls instead of just dialing multiple extensions with the &amp; operator?

Any technology recommendations or examples for integrating VoIP with open source chat platforms like Jitsi, Mumble, etc.?

Does anyone have SIP setup on their domain? Are there many spam calls these days?

One of my first job, in 201, was to create a spam faxing machine based on Asterisk. Once the initial setup was completed (T38 is a bit tricky to tune), it was very effective, it sent more than one milion of faxes every year.