New: Comfortable and robust VOIP without PBX in AREDN networks
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- Опубликовано: 11 авг 2023
- When it comes to emergency communications, it must be solid, reliable, and scalable. In HAM radio, we often use AREDN networks for that purpose. Because they run on 2.4 or 5.8GHz, their performance is high and allows to build mesh networks for flexibility and reliability. Today, we will add the next level: Redundant Voice over IP. Be with me when one of my dreams comes true!
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I love the guy with the Swiss accent
:-)
Grüezi from Florida
Grüezi back to Florida!
This is a fun project!
By the way, you don't need a router or remote phone books to do this with Yealink phones. Yealink has the ability to use custom dial plans just like Asterisk. I have a couple of the old Yealink T32's setup on their own POE switch and physically separated from any other network connection. My kids use them to play phone between rooms. Depending on what they dial they may get a voicemail, intercom, or regular calling.
Interesting. So your dial plans keep the ip addresses of the phones?
I love this. I've just setup HamsOverIP and the Hamshack Hotline on my SIP phone. The HSHL has trunks to AREDN so we can talk freely to each other too :)
Cool. I do not know these two protocols. They seem to be more used in the US.
Merci Andreas für das Webinar, deine super Unterstützung, service après-vente ;-) 73's
Gern geschehen!
Very interesting!
Thank you!
Hi, I can help you with the cross compiling. I have long experience developing software for that router and others qualcom mips processors. Also I ‘m voip expert. Your mesh voip network it’s interesting. In the real world we use dns servers to solve that problem, by using an special kind of dns record called SRV with allow us set an ordered list of sip servers.
Thank you for your offer. In the meantime we managed to get the thing compiled and packaged for AREDN. But we are always interested in people with knowhow interested in the topic. Maybe you send me a note? You find my email on QRZ.com.
This is something I am going to look into with Cisco phones and have some fun.
Maybe you can document what you needed to do?
@@HB9BLA Have to figure out if can make it work first and how to do that. Spent to much time running PBX's.
I love this idea. It looks like someone still needs to manually maintain the phonebook that will be converted to XML and distributed. I must have missed how the distribution from hAP to hAP works automatically.
They are regularly and automatically downloaded from a server during good times. At bad times when the server is lost, they just stay there.
good fun
Agreed!
Reminds me of my Voip box back in 2004, Andreas. Only a chinese coin this week, his gold is hard to find.
I assume this is why it is so expensive ;-)
I think Mikrotik can run containers now, maybe this would help?
These small routers and antennas only have 64MB of flash. So containers probably would be overkill. But we want to use a Raspberry in the mesh for that purpose. However, this is then a single point of failure and cannot be used for important stuff.
Hi. I have been designing VoIP systems for many years. I think you'll be better off if you introduce federated proxies. A proxy properly configured on each node will take care of routing calls internally within the node. If the callee does not belong to the node, the call will be forwarded to the proxy handling particular callee. Of course audio will still flow directly between phones so no delay or distortions will be introduced. You will probably need a better SIP proxy such as Kamailio or Opensips. It will still work fine in split-brain scenario when a node is disconnected from the mesh. Have a think and let me know if you need any help.
Of course, I am always interested in better solutions! I am no VOIP specialist. This is why I do not understand how this would work (I also do not understand the difference between a PBX and a proxy for small networks). We generally only have one or 2 phones per hap router. Would such a router be the "node"? And how would the decentralized phonebook be handled? The IP address of the phones will change over time because the mesh is not stable (AREDN does all this management). And what would be the advantages? I do not post my email on RUclips, but I am on QRZ or on Twitter? I would be interested in learning your approach.
@@HB9BLA one of the features of SIP proxies (and not only proxies) is the registrar database. When the phone registers to it, the registrar stores client information (such as IP address / port of the client) and can use it to route the call when needed. For example, you could register as hb9bla (at) node1, and if anyone calls that SIP URI, the proxy will translate it to the IP address and send it to you. Practically you will never need to generate dynamic phonebooks because your SIP address will never change.
Let me compie the aredn client for x86-64 and will prepare a real live demo with it.
Are configurations for Cisco phone available?
Does the phonebook use hash or blockchain validation?
Nothing like that.
q1 - wouldn't it be better using IPv6 (if the phones can make us of that), since it is much easier to handle when it boils down to IP-Address-handling (eventough the address as such is more complicated), the non-publicly-routed FE80-address can be used within the mesh, and it normally will not change (for the MAC-ID being part of the fixed address),
q2 - is there a "AREDN for "non licensed but interested people""? (like: pure wlan-meshes trying the same, or even: using CB for the same venture?) OR:
q3 - since i have lots of trouble learning the skills needed for the hb3 and only have hostile clubs in my area (all of my good school-friends are members there - and still didn't grow up): what would be your advice for beginners?
1. AREDN (like many others) does not support IPV6
2. You could use regular ISM frequencies for your mesh. A few channels are shared by HAM radio and ISM. However, I do not know the legal implications.
thank you @@HB9BLA
AREDN does not function as a network. I am located in San Francisco, where one of the largest AREDN networks exists, and I have conducted extensive testing on the network.
To implement FOSS, AREDN OpenWRT employs CSMA (802.11ac ad-hoc mode) instead of proprietary TDMA signaling. CSMA is effective for a single user, but its performance deteriorates rapidly when multiple users are present on a network. This is why network demonstrations are never shown, only link demonstrations.
If you recall ALOHAnet, it used CSMA and served as the motivation for the development of improved demand access schemes for radio networks, such as TDMA. In overload conditions, channel utilization in CSMA can drop to zero.
Jitter increases with channel contention, rendering VOIP unusable within a network.
Please refrain from spreading misinformation. Is there anything else you would like me to help with? 😊
This is an unhelpful and misleading comment.
The barest skim of IEEE 802.11 would inform you all wifi devices (at least all that adhere to 802.11 specs) utilize carrier sense multiple access with collision avoidance (CSMA/CA). As CSMA/CA is implemented as part of the physical layer, there is nothing AREDN, OpenWRT, or any other firmware can do to escape utilizing CSMA/CA. There is nothing proprietary about time division multiple access (TDMA), it simply is not part of the 802.11 spec. Stating any of this is done "To implement FOSS" is completely nonsensical in any context.
If you have done extensive testing of the network, you must have documented and repeatable results. Can you share these results?
@@shotgunmooseI have posted results posted in HaD but they were taken down when linked here. I am able to get AREDN to lock up and reduce bandwidth to zero in typical operation. Take a look at IEEE 802.11e-2005 - that fixes problems with 802.11 ad-hoc mode, like lack of QoS. TDMA avoids a boatloads of ALOAHAnet problems. There's no good reason to use ad-hoc mode when TDMA is available.