I am looking for an adapter capable to convert multiple RS-485 ports to LoRaWAN. There are several in the market, but the majority offer only One (1) port. I need to connect two different field RTU devices remote device independently.
I have seen several vendors offering one (1) single port. Although I can use one port (Master)to connect several field RTU devices remote device, I need to modify the address for each one (slave) on each RTU device. Sometimes, there is no access to the configuration of these units, since a vendor does not allow.
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I'm controlling a couple of i2c slaves (PCA9505) using the I2C module of National Instruments LabVIEW. I need to improve the project so I can add new slaves to the bus without manually adjusting the address of the slaves. (pins A0 A1 A2 on PCA9505). As a consequence, hardcoding the addresses is not a option.
Also, PCA has only 3 pins, so I can add max 8 slaves. I need to go for 16.
How can I give the slaves dynamic addresses?
How can I expand to 16 slaves?
Thanks for your help!
PS: PCA9505 = 40-bit parallel input/output (I/O) port expansion for I2C-bus applications
Personally, I like to use a configuration file to set the parameters needed for an application. In this case, it would define the addresses for each device you need to talk to. Arrays also help a lot here for expandability.
As far as extending your I2C bus, you can use something like a PCA9544A I2C mux. This chip allows for controlling 4 different buses. You will have to control the mux as part of your addressing of the individual chips.
From a brief read of the PCA9505 datasheet it doesn't look as if there's any way of selecting the device's address other than using those address lines. You are either going to have to multiplex the bus as crossrulz suggests, so that each device is at the same address on a different I2C bus, and add code to control which one you talk to at any one time, or add some hardware that sets the address lines differently for each device.
Will the devices be added to the bus via some sort of connector? If so could you include the address lines in the connector, so that each connector on the master adapter sets a different address for the device that's plugged in to it? Or if you connect devices in a daisy chain, each device could add 1 to the address on its upstream connector and output the new address on its downstream connector?
I encounter a problem when I'm trying to use a Ubuntu laptop computer to connect to two Sony A6000 through WiFi in the same time.
On that laptop there are two WiFi adapters, one is embedded (say, Wifi-A) and the other is an Asus USB-N13 ProN USB
dongle (say, Wifi-B).
Using Sony Camera Remote API I can successfully control these two A6000 (say, A6000-A & A6000-B)
through these two Wifi adapters, provided only one camera is connected a time. For example, either using Wifi-A to control to
A6000-A or A6000-B, or using Wifi-B to control A6000-A or A6000-B, will work.
I think this proves both the cameras and Wifi adapters hardware function correctly, and any combination of Wifi adapter and
camera works fine, too.
However, it will be different if I try to connect and control both cameras in the same time. The connection to both cameras
is OK (though sometimes not very smooth), but controlling them is not.
Here is a list of the experiment steps:
Wifi-B connects to A6000-B. Then Wifi-B gets IP address 192.168.122.166
Wifi-A connects to A6000-A. Then Wifi-A gets IP address 192.168.122.165
Send M-SEARCH request to Wifi-A and get response from A6000-A. Then get
Device Description XML file from A6000-A successfully.
Send M-SEARCH request to Wifi-B and never get response
I'm wondering if such configuration (One laptop + two Wifi adapter + two A6000) violates any design consideration
of A6000? For example, because both cameras use the same IP address 192.168.122.1 for themselves?
I'll appreciate if any one could comment on this issue.
Thanks in advance!
Xavier
After doing experiments for many days, I'm sure this symptom is caused by the IP address conflict (both of them are 192.168.122.1) of these two A6000.
As I can't find a way to change them, my optimal solution under such configuration (one laptop + two Wifi adapter + two A6000) is to make use of Linux Network Namespaces. Moving Wifi-B to a new Network Namespace does solve this issue, because every Network Namespace has its own network stack. Now I can freely access both of these A6000 from my laptop.
I hope this answer is helpful to anyone who encounters a similar problem.
There isn't any design consideration that would block this from working. Unfortunately we cannot offer any other troubleshooting suggestions.
I newly wrote a simple chat application, but I didn't really understand the background of ICE Candidates.
When the peer create a connection they get ICE Candidates and they exchange them and set
them finally to the peerconnection.
So my question is, where do the ICE Candidates come from and how are they used and are they all really used ?
I have noticed that my colleague got less candidates when he executes the application on his machine, what could be the reason for different amount of Candidates ?
the answer from #Ichigo is correct, but it is a litte bit bigger. Every ICE contains 'a node' of your network, until it has reached the outside. By this you send these ICE's to the other peer, so they know through what connection points they can reach you.
See it as a large building: one is in the building, and needs to tell the other (who is not familiar) how to walk through it. Same here, if I have a lot of network devices, the incoming connection somehow needs to find the right way to my computer.
By providing all nodes, the RTC connection finds the shortest route itself. So when you would connect to the computer next to you, which is connected to the same router/switch/whatever, it uses all ICE's and determine the shortest, and that is directly through that point. That your collegue got less ICE candidates has to do with the ammount of devices it has to go through.
Please note that every network adapter inside your computer which has an IP adress (I have a vEthernet switch from hyper-v) it also creates an ICE for it.
ICE stands for Interactive Connectivity Establishment , its a techniques used in NAT( network address translator ) for establishing communication for VOIP, peer-peer, instant-messaging, and other kind of interactive media.
Typically ice candidate provides the information about the ipaddress and port from where the data is going to be exchanged.
It's format is something like follows
a=candidate:1 1 UDP 2130706431 192.168.1.102 1816 typ host
here UDP specifies the protocol to be used, the typ host specifies which type of ice candidates it is, host means the candidates is generated within the firewall.
If you use wireshark to monitor the traffic then you can see the ports that are used for data transfer are same as the one present in ice-candidates.
Another type is relay , which denotes this candidates can be used when communication is to be done outside the firewall.
It may contain more information depending on browser you are using.
Many time i have seen 8-12 ice-candidates are generated by browser.
Ichigo has a good answer, but doesn't emphasise how each candidate is used. I think MarijnS95's answer is plain wrong:
Every ICE contains 'a node' of your network, until it has reached the outside
By providing all nodes, the RTC connection finds the shortest route itself.
First, he means ICE candidate, but that part is fine. Maybe I'm misinterpreting him, but by saying 'until it has reached the outside', he makes it seem like a client (the initiating peer) is the inner most layer of an onion, and suggests the ICE candidate helps you peel the layers until you get to the 'internet', where can get to the responding peer, perhaps peeling another onion to get to it. This is just not true. If an initiating peer fails to reach a responding peer through the transport address, it discards this candidate and will try a different candidate. It does not store any nodes anywhere in the candidate. The ICE candidates are generated before any communication with the responding peer. An ice candidate does not help you peel the proverbial NAT onion. Also regarding the second quote I made from his answer, he makes it seem like ICE is used in a shortest path algorithm, where 'shortest' does not show up in the ICE RFC at all.
From RFC8445 terminology list:
ICE allows the agents to discover enough information
about their topologies to potentially find one or more paths by which
they can establish a data session.
The purpose of ICE is to discover which pairs of addresses will work. The way that ICE does this is to systematically try all possible pairs (in a carefully sorted order) until it finds one or more that work.
Candidate, Candidate Information: A transport address that is a
potential point of contact for receipt of data. Candidates also
have properties -- their type (server reflexive, relayed, or
host), priority, foundation, and base.
Transport Address: The combination of an IP address and the
transport protocol (such as UDP or TCP) port.
So there you have it, (ICE) Candidate was defined (an IP address and port that could potentially be an address that receives data, which might not work), and the selection process was explained (the first transport address pair that works). Note, it is not a list of nodes or onion peels.
Different users may have different ice candidates because of the process of "gathering candidates". There are different types of candidates, and some are obtained from the local interface. If you have an extra virtual interface on your device, then an extra ICE will be generated (I did not test this!). If you want to know how ICE candidates are 'gathered', read the 2.1. Gathering Candidates
I've read a lot about WebRTC, but there's one question that still remains. I hope you can help me with that:
Does WebRTC allow me to create a one-to-many connection? I don't mean "being able to have multiple connections to different computers", I really talk about having one connection that multicasts its data to multiple endpoints without the need to "upload" the data once for each endpoint. Will it be possible to send one single package to the web, that, when it reaches the web, magically splits itself into multiple packages with different targets?
I hope you get what I'm looking for :)
Until now, I've only seen one-to-one connections, or solutions that have one connection to a central server that does the multicast for them (which usually results in twice the ping).
But to me, one-to-one connections don't seem to be really useful (due to low upload-bandwith of clients), and solutions with a central server are also possible without WebRTC (using WebSockets), so the only real use case for WebRTC would be one-to-many connections.
So.. is this something that will be possible in the future? Or is it already possible today?
Three things:
IP multicast in the Internet is not possible at the moment (multicast addresses are not routed by ISPs)
WebRTC fits many use cases beyond one-to-many communication, just have a look at this document: https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-use-cases-and-requirements-06
WebRTC connections between browsers are always encrypted (using SRTP for A/V data and DTLS for generic data) and the encryption parameters (session keys etc.) are negotiated for every connection separately. How would you do that in a multicast environment (think of it as a distribution tree)?
So no, WebRTC cannot be used with IP multicast.
I would answer "It doesn't for now", because as a programmer, I can tell you, that there are number of ways browser devs to make it work if we (users) insist on it's importance. But how ? Since there's encryption, they could allow sharing of the session's encryption keys to the group of 'registered' (multicast) users. But how ? Well, Web was created for sharing. The most obvious way is through web server mediation and JS WebRTC API function (to load the user keys). Since multicast is most often used for efficient video distribution, you have a RTP/SRTP video server. The web server can coexist at the same machine. If they decide to extend it to web browsers - then just the "server" role can be done by the Web browser who created the multicast stream (the sender). The clients need to know who is it.
Again: In December 2013, this is still not possible. And multicasts are allowed on the Internet only in:
some experimental WAN nets
some internet+video ISP nets
LANs (when enabled at switch level, cheap switches transmit it to all ports). But you can be an ISP, researcher or LAN user, so it's necessary.
As you know we can use our ADSL modems to send/receive data with high bandwidth through our phone line to/from ISP servers. I'm interested to write a program to use this capability of ADSL modem to send/receive files directly between computers (like Hyper Terminal application which was used for 56k serial modems). Is it possible? Also I want to know does it need governmental support or not? :)
Best Regards
While this is definitly possible (at least with the Alcatel ADSL modems), it is not worth the hassle. Some random points - please mind that this is written from a european POV, things might be a little different in the USA.
Carriers are not amused, if you change the configuration of your modem. Some will call it "hacking" and disconnect you.
It is not possible, to use the modem for both (Internet access and file transfer) at the same time.
It won't work between modems on different carrier networks or even in different LM segments of the same carrier - the ADSL signal is terminated at the multiplexer in your carrier's LM switch. It does however work between modems attached to the same multiplexer.
For your problem, the solution ist to use a TCP connection instead of a fake serial - even HyperTerminal can do that AFAIR