I'm currently using Redis in an IoT application to receive a stream of data from an acquisition board; all other communications between the PC and the board is based on Modbus/TPC protocol.
A colleague of mine has recently advanced the proposal to completely remove Modbus, and use Redis for all communications instead.
Supposedly this would require a mixture of variables exchange and PUB/SUB signals.
While the idea is attractive, I was just wondering if someone has already done some research in this direction.
Modbus is a widely used protocol to communicate between industrial devices on one side and computers / gateways on the other side. The device is the server, the computer is the client. Sensor data is polled, changes are pushed.
Redis provides a protocol RESP https://redis.io/topics/protocol between REDIS clients and the Redis server. The devices would then be clients, and the computer the server.
Replacing modbus with RESP would thus invert the client/server relationship.
While there are advantages (better typed data transfer) its uncommon the select a RESP in that area. MQTT or so would be more common.
Related
I would like to create an audio (mic) and video (camera) chat room with 12 people using webRTC. I understand signalling and the need for external services like ICE & STUN to help peers connect with each other.
But I don't want to use a full mesh architecture where everyone connects to everyone else because it is less efficient. I don't want to use expensive TURN relay services. I want the swarm to propagate the streams automatically so that if a direct connection isn't possible, the network routes the stream packets via peers automatically using encapsulation.
I don't want to use star architecture because I don't want a bottleneck peer.
I would like a peer to connect to maybe 2 or 3 other peers (max) and broadcast their media across the network without worrying about the relaying between peers. The routing would obviously need to be controlled by some service but I can't see that it's possible to do the stream encapsulation with RTCPeerConnection. Since WebRTC only allows for a RTCPeerConnection object (1 peer to 1 peer) and a peer would need to distinguish where the incoming stream is coming from and whether it needs to be relayed to another peer.
Is there a technology that extends WebRTC to allow for this more bandwidth efficient architecture?
I have a configuration where 3 applications run on 3 different Virtual Machine's and they must communicate via DDS i.e. RTPS protocol.
The configuration is as follows :
ROS2 based ADAS functions
Simulation Tool
Python/Tensorflow based machine learning functions
All 3 need to be on different VMs.
It is not possible at our end to allow multicasting for the MS AZURE VM and our network.
Here are some questions :
Is it still possible to communicate via DDS ?
If yes, through UNICAST i.e. peer to peer connection ?
Is using DDS communication beneficial in this case if i already have the option of basic UDP socket programming ?
Could you think of any restrictions/ further problems in using DDS for such a configuration ?
Is it still possible to communicate via DDS ?
Yes, it is. Out of the box, DDS Participants only use multicast for discovering other DDS Participants, at startup. This discovery mechanism can be configured in several ways. For a an explanation on how to achieve that, see this RTI Community Knowledge Base article: Configure RTI Connext DDS to not use Multicast.
If yes, through UNICAST i.e. peer to peer connection ?
Yes, with the no-multicast setup, all communications are done over UDP unicast, peer to peer, connectionless.
Is using DDS communication beneficial in this case if i already have the option of basic UDP socket programming ?
Not being able to use multicast does not remove any of the DDS advantages when comparing it to UDP. When using DDS, the transport/discovery configuration is typically invisible to the application and all Publish/Subscribe concepts remain unchanged.
If you are asking about the advantages of using DDS versus UDP, I think that warrants a new question on itself. The answer will be quite extensive :-)
Could you think of any restrictions/ further problems in using DDS for such a configuration ?
With this configuration, your configuration settings will be dependent on the network that you are running on. This means that migration to a different network might need reconfiguration, for example providing different host names or IP addresses. This is inconvenient, but not hard.
Since your environment is restricting the use of multicast, I would not be surprised if there are more restrictions that you have not mentioned or discovered. For example, do you know anything about firewalls or network bandwidth restrictions? Again, DDS can be configured to deal with such things, but you need to be aware of them first.
I don't have a specific query here ; just need some design guidelines.
I came across this article on Node.js , MQTT and Websockets.
I guess we can achieve similar purpose using Node/Java + ActiveMQ + Websockets. My query is how to select between MQ and MQTT ? Can I safely use an "open" server like mosquitto in a medium-large scale project, compared to ActiveMQ ?
This article has had some insight, and it seems like I should use both MQ and MQTT, as MQTT may possibly help if I get lightweight clients in future.
Thanks !
Adding to what Shashi has said, these have different capabilities and use cases.
MQTT defines a standard wire protocol for pub/sub and, as Shashi noted, is designed for very lightweight environments. As such it has a very minimal wire format, a few basic qualities of service and a basic feature set.
Traditional message queueing systems on the other hand are generally proprietary (although AMQP aims to change that), cover both point-to-point and pub/sub, offer many qualities of service and tend to have a more heavyweight wire format, although this exists to support enhanced feature sets such as reply-to addressing, protocol conversion, etc.
A good example of MQTT would be where you have endpoints in phones, tablets and set-top boxes. These have minimal horsepower, memory and system resources. Typically connections from these either stay MQTT and they talk amongst themselves, or they are bridged to an enterprise-class MQ where they can intercommunicate with back-end applications. For example, an MQTT-based chat client might talk directly to another through the MQTT broker. Alternatively, an MQTT-based content-delivery system would bridge to an enterprise messaging network which hosted the ads and other content to be delivered to apps running on phones and tablets. The enterprise back-end would manage all the statistics of ad delivery and views upon which billings are based and the MQTT leg allows the content to be pushed with minimal battery or horsepower consumption on the end-user device.
So MQTT is used for embedded systems and end-user devices where power, bandwidth and network stability are issues. This is often in combination with traditional MQ messaging, although I haven't ever seen MQTT used as the exclusive transport for traditional messaging applications. Presumably, this is because MQTT lacks some of the more robust features such as message correlation, reply-to addressing and point-to-point addressing that have been core to messaging for 20 years.
MQTT protocol is suited for small devices like sensors, mobile phones etc that have small memory footprint. These devices are typically located in a brittle network and typically have low computing power.
These devices connect to an organizations back-end network via MQTT protocol for sending and receiving messages. For example a temperature sensor in an oil pipeline would collect temperature of the oil flowing through the pipe and send it to the control center. In response a command message could be sent over MQTT to another device to reduce/stop the flow of oil through that pipe.
WebSphere MQ has the capability to send/receive messages to/from the MQTT devices. So if you plan to implement a messaging based solution that involves devices & sensors, you can consider MQ and MQTT.
HTH
As already discussed, MQTT defines an applicative wire protocol (i.e. how the information is organized and then serialized, before to be transferred).
Mosquitto, or whatever else MQTT broker, is just an implementation of the Hub and Spoke Integration Pattern, just like JMS and AMQP based brokers, the difference consists in the wire protocol at transport level: AMQP defines a standardized transport wire protocol, instead JMS brokers like ActiveMQ defines their own proprietary format, namely the OpenWire. Of course, not standard implementations, like Mosquitto, implement proprietary wire transport protocol (this impacts interoperability, but can be a better choice in terms of perfomances).
Back to the question. Brokers like Mosquitto can be used in real scenarios, according to your needs in terms of scalability and reliability: normally, clustering is needed to assure i. Availability, ii. Reliability and iii. Scalability. Brokers thought for PAN (Private Area Netorks), normally do not provide OTB (Out of The Box) such features - ActiveMQ provides that.
Concluding, it's up to your requirements to pick for you the best solution.
What is the most simple and straightforward approach for serverless P2P UDP chat in Boost Asio? The chat will work across the internet. There are ready UDP examples but they all maintain client-server approach!
I'm assuming that by serverless you mean a P2P network without a central control server.
IMO your question has little to do with boost-asio. asio is a cross-platform network library. You seem to be asking more of a network engineering type question and asio is just one of the tools you can use for implementation.
The examples are client-server in that the example applications possibly fall under the client-server architecture. However the socket code (or asio usage) used to send and receive messages will look the same irrespective of client-server or P2P applications i.e. you send a message to an address and you receive messages on a specified port. The differences will come in at the protocol layer, but this has nothing to do with asio per se.
The following may or may not be of interest to you: there is no simple way IMO: users located behind firewalls and NAT means that you need to use techniques such as STUN, TURN and ICE to resolve addresses or in the worst case relay data. All these designs require a server.
Can anyone tell be where to use the UDP protocol except live streaming of music/video? What are default usecases for UDP?
UDP is also good for broadcast, such as service discovery - finding that newly plugged in printer.
Also of note is that broadcast is anonymous, you don't need to specify target hosts, as such it can form the foundation of a convenient plug-and-play or high-availability network.
UDP is stateless and is good for applications that have large numbers of clients connecting to a server such as time servers or DNS. The fact that no connection has to established and maintained reduces the memory required by the server. There is no handshaking involved and so this reduces the traffic on the network. On the downside, if the information transferred requires multiple packets there is no transmission control to ensure that all packets arrive and in the correct order - but in games packets lost are probably better than late or disordered.
Anything else where you need performance but can survive if a packet gets lost along the way. Multiplayer games come to mind, for example.
A very common use case is DNS, since the overhead of creating a TCP connection would by far outweight the actual payload.
Additional use cases are NTP (network time service) and most video games.
I use UDP to add chat capabilities to our applications. No need to create a server. It is also useful to dispatch events to all users of our applications.