I want to use TLS with stm32f4 for the communication safe for the modbus-tcp, but I dont want to use RTOS, how can I do that ? Is that possible? I am using lwip raw api, lwip documentation says it is possible but what is the method ?
Related
I have program which performs successful transfer of data between client(washing machine) and server(HawkBit) by using FreeRTOS+TCP in which we have sockets from FreeRTOS only but now I want to implement TLS over it for which I am using mbedTLS in which we again have function for TCP connections, functions for socket opening closing, every function which we have in FreeRTOS again now I don't know what should I do...!is there any possibility of using the socket from FreeRTOS+TCP and just make CA certification from mbedTLS on the top of it! or I need to implement again everything with mbedTLS what I have implemented before with freeRTOS!
This is using FreeRTOS+TCP through a TLS abstraction https://github.com/aws/amazon-freertos/blob/master/libraries/abstractions/secure_sockets/freertos_plus_tcp/iot_secure_sockets.c - and this is the implementation of the abstraction layer for mbedTLS https://github.com/aws/amazon-freertos/blob/master/libraries/freertos_plus/standard/tls/src/iot_tls.c
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'm building a spring-websocket application that currently uses RabbitMQ as a message broker via the STOMP protocol. The rest of our organization mostly uses IBM Websphere MQ as a message broker, so we'd like to convert it away from RabbitMQ. However Websphere MQ doesn't support the STOMP protocol, which is spring-websocket's default. MQTT seems like the easiest supported protocol to use instead. Ideally our front-end web clients will continue to use STOMP, but I'm also OK with migrating them to MQTT if needed.
What classes do I need to overwrite to make spring-websocket interface with the broker via MQTT instead of STOMP? This article provides some general guidance that I should extend AbstractMessageBrokerConfiguration, but I'm unclear where to begin.
Currently I'm using the standard configuration methods: registry.enableStompBrokerRelay and registerStompEndpoints in AbstractWebSocketMessageBrokerConfigurer
Ryan has some good pointers.
The main work is going to be creating a replacement for StompBrokerRelayMessageHandler with an MqttBrokerMessageHandler that not only talks to an MQTT broker but also adapts client STOMP frames to MQTT and vice versa. The protocols are similar enough that it may be possible to find common ground but you won't know until you try.
Note that we did have plans for for MQTT support https://jira.spring.io/browse/SPR-12581 but the key issue was that SockJS which is required over the Web for fallback support does not support binary messages.
Here's my stab at this after reviewing the spring-websocket source code:
Change WebSocketConfig:
Remove #EnableWebSocketMessageBroker
Add new annotation: #EnableMqttWebSocketMessageBroker
Create MqttBrokerMessageHandler that extends AbstractBrokerMessageHandler -- suggest we copy and edit StompBrokerRelayMessageHandler
Create a new class that EnableMqttWebSocketMessageBroker imports: DelegatingMqttWebSocketMessageBrokerConfiguration
DelegatingMqttWebSocketMessageBrokerConfiguration extends AbstractMessageBrokerConfiguration directly and routes to MqttBrokerMessageHandler
Add this to server.xml on WebSphere Liberty:
<feature>websocket-1.1</feature>
I'm trying to setup a call between webRTC based client (olympus) and a standard one (x-lite i.e.).
The call is failing (480). I believe it is because of SDP negotiation failed.
Currently I use standard telestax mediaserver setup.
Can restcomm be configured in a way, that it transcodes the stream (and modifies codec negotiation), so webRTC based clients can call the standard sip ones ?
Thank you very much in advance.
Hubert
I'm going to implement Java VoiP server to work with WebRtc. Implementation of browser p2p connection is really straightforward. Server to client connection is slightly more tricky.
After a quick look at RFC I wrote down what should be done to make Java server as browser. Kindly help me to complete list below.
Implement STUN server. Server should be abke to respond binding
request and keep-alive pings.
Implement DTLS protocol along with DTLS handshake. After the DTLS
handshake shared secret will be used as keying material within SRTP
and SRTCP.
Support multiplexing of SRTP and SRTCP stream. SRTP and SRTCP use
same port to adress NAT issue.
Not sure whether should I implement SRTCP. I believe connection will
not be broken, if server does not send SRTCP reports to client.
Decode SRTP stream to RTP.
Questions:
Is there anything else which should be done on server-side ?
How webRtc handles SRTCP reports ? Does it adjust sample rate/bit
rate depends on SRTCP report?
WebRtc claims that following issues will be addressed:
packet loss concealment
echo cancellation
bandwidth adaptivity
dynamic jitter buffering
automatic gain control
noise reduction and suppression
Is is webRtc internals or codec(Opus) internals? Do I need to do anything on server side to handle this issues, for example variable bitrate etc ?
The first step would be to implement Interactive Connectivity Establishement (RFC 5245). Whether you make use of a STUN/TURN server or not is irrelevant, your code needs to issue connectivity checks (which use STUN messages) to the browser and respond to the brower's connectivity checks. ICE is a fairly complex state machine, but it's doable.
You don't have to reinvent the wheel. STUN / TURN servers are external components. Use as they are. WebRTC source code is available which you can use in your application code and call the related methods.
Pls. refer to similar post - Server as WebRTC data channel peer