I found this Answer RSMB MQTT-SN & Bluetooth, but I am not sure if this is really the correct answer at all.
So a second Question - I am new to Stackoverflow so I cannot comment directly.
...
Are you sure that a forwarder is really needed here? I read the MQTT-SN spec and for me it looks like MQTT-SN is for UDP and UDP is connectionless. So I think it is possible so simulate UDP over serial for one point to point connection.
So why not...
mqtt-sn client---serial-->> any radio <<--serial---mqtt-sn serial brigdge
And on the MQTT-SN serial bridge side I can also run a Gateway which connects to a real MQTT broker of my choice.
I read that out from figure 1 in the specs. I do not clearly understand what´s the benefit of a forwarder is? And when should someone use it and so on ...
thanks. Mathias
The forwarder encapsulates the address on the radio network of the client, so the broker can reply to the right client when there is more than one client.
Related
For learning purposes I'm implementing TCP (for now just RFC 793) but I have no
idea how to test it. Most TUN/TAP stuff on the internet are out of date (e.g.
Linux API calls no longer work) and just doesn't explain enough. In addition, I
feel like a creating a device and forwarding packages etc. are not the best way
for learning purposes. For example, I'd rather only override socket(),
listen(), connect(), accept(), send(), recv() etc. in a program rather
than forwarding all ethernet traffic to a device/program that does the
bookeeping for the whole system rather than for a single program.
I'm wondering if this is possible. If not, I'd like to know the simplest way to
test a TCP implementation on Linux.
Because I'm following RFC 793, it'd be great if I could have an IP (Internet
Protocol as mentioned in the RFC) API in my application. Is this possible or do
I have to mess with TUN/TAP stuff?
Thanks..
If we talk about research I strongly recommend you read Engineering with Logic: Rigorous Test-Oracle Specification and
Validation for TCP/IP and the Sockets API
It contains section about testing TCP/IP implementation:
"EXPERIMENTAL VALIDATION: TESTING INFRASTRUCTURE"
You could try setting up two peers, one using a RAW socket and the other a TCP socket.
If they can communicate and packets are really delivered/recovered the same way TCP does, you know that your custom implementation is successful.
C sockets
C RAW sockets
C TCP implementation
All you need is to intercept all tcp packets with bits (syn, ack, fin, etc.) your application has sent and to see if it works properly:
It could simply be done with wireshark or other sniffer. When testing you will see all tcp packets with bits you've sent.
In order you want to see linux system calls which your application are calling, you can use GDB or any other debugger.
I am working with an IP camera based on onvif standards. The scenario is the following one:
The RTP port and HTTP port are 22554 and 22280 respectively. I have no problem with this, I reroute the traffic from A to B in those ports to reach the camera. In fact, the RTP session is correctly established.
The problem comes in the SETUP message. Here A (which starts the session), establishes a new port (client port) for the UDP connection for video interchange. The same way, the camera also sets a new port (server port). As the server ports are not redirected in B, I am not able to receive anything.
I read that there is a possibility to used interleaved mode in order to use the same port used for RTSP messages (22554 in this case) for the video packets. But I do not know how to do this. For example, VLC tries the first option (new client-server ports) and as it does not work it successfuly setups another rtsp session with interleave mode.
I started working with this code (http://bit.ly/1Xvwqx9), which is based on Onvif libraries. But I cannot find anything regarding this aspect of interleaving.
Anyone can give me a hint?
Thanks and kind regards.
Finally I found the solution. It is important to set these aspects:
protocol = TransportProtocol.rtsp
MediaStreamInfo.Transport transp = MediaStreamInfo.Transport.Tcp;
I'm not quite sure how the firewall can tell what transportation is being passed - TCP\UDP?
also - when I have statefull VS stateless FW - I know that there is the difference when using TCP, but what about UDP?
thanks alot! :)
Have a look at https://en.wikipedia.org/wiki/IPv4#Header and see the Protocol field.
The header of the packets are completely different, TCP is much bigger for example. A stateful firewall needs to intercept the TCP headers to map the packets to its state table, but also stateless firewalls sometimes have techniques implemented to recognize valid TCP or UDP headers. Most home routers (broadband/wireless) make use of this when you are using port forwarding to distinguish between both protocol versions.
Assuming that you have determined that for a given niche case, neither TCP or UDP are ideal, how would you go about writing your own IP based protocol?
For example, if you're developing on Linux, where would you look in the kernel to "hook" your protocol in?
Where would you start?
You can do this through a kernel module. I would start by reading how arp works for example. That is a simpler protocol since userspace doesn't send packets out with it directly.
The entry point for creating a new network protocol is dev_add_pack, and the code for arp can be found here.
If your protocol can be implemented directly on top of IP, then it can also be implemented wrapped in UDP packets - and the latter has the advantage that it'll pass through existing NAT devices and firewalls that would simply drop your custom protocol.
Read up on UNIX sockets and networking. It's not so much 'hooking' into the kernel, as it is opening a socket and sending your binary data over that.
I need an EXPERT opinion please, and sorry if my question itself is a confused question.
I was reading around about structure of VOIP applications (Client/Server). And mostly UDP is recommended for voice streams. I also checked some voicechat applications like paltalk and inspeak and their sites mention they use udp voice stream which dont seem correct for below reasons.
I examined the traffic/ports used by paltalk and inspeak. They have UDP and TCP ports open and using a packet sniffer i can see there is not much UDP communication but mostly it is the TCP communication going on.
Also as far as i know, In UDP Protocol server can not send data to a client behind NAT (DSL Router). And "UDP Braodcast" is not an option for "internet" based voice chat applications. THATS WHY YAHOO HAVE MENTIONED in their documentation that yahoo messenger switch to tcp if udp communication is not possible.
So my question is ....
Am i understanding something wrong in my above statements ?
If UDP is not possible then those chat applications use TCP Stream for voice ?
Since i have experienced that TCP voice streams create delay, No voice breaking but Delay in voice, so what should be the best structure for a voice chat server/client communication ?
So far i think that , if Client send data as udp packets to server and server distribute the packets to clients over TCP streams, is this a proper solution ? I mean is this what commercial voicechat applications do ?
Thanks your answer will help me and a lot of other programmers .
JF
UDP has less overhead (in terms of total packet size), so you can squeeze more audio into the channel's bandwidth.
UDP is also unreliable - packets sent may never be received or could be received out of order - which is actually OK for voice applications, since you can tolerate some loss of signal quality and keep going. a small amount of lost packets can be tolerated (as opposed to downloading a file, where every byte counts).
can you use TCP? sure, why not... it's slightly more overhead, but that may not matter.
SIP is a voice/media standard that supports UDP and TCP. most deployments use UDP because of the lower overhead.
The Skype protocol prefers UDP where possible, and falls back to TCP.
in SIP situations, the NAT problem is solved by using a nat keep-alive packet (any request/response data) to keep the channel up and open, and by exploiting the fact that most NATs will accept replies on the same source port the connection was opened from... this isn't foolproof, and often requires a proxy server mediating the connection between 2 nat'd peers, but it's used in many deployments.
STUN, TURN, and ICE are additional methods that help with NAT scenarios, and especially in p2p (serverless) situations.
info regarding NAT issues and media:
http://www.voip-info.org/wiki/view/NAT+and+VOIP
http://en.wikipedia.org/wiki/UDP_hole_punching
http://www.h-online.com/security/features/How-Skype-Co-get-round-firewalls-747197.html
if you're implementing a voice service of some kind, a system like FreeSwitch provides most of the tools you need to deliver media to distributed clients:
http://www.freeswitch.org/
I see the question is 3 years overdue, but I see no answer accepted, so I'll take a shot at it
1- your statements are correct
2- correct, TCP or UDP can be used for audio stream.
3- Combining tcp and udp for the audio stream is not useful. If UDP is working for transmission to the server, it will work for reception, that's how all NAT firewalls work, i.e they send datagram received from internal host to remote host after they change the ip header to make the packet seem coming from them, and when they receive response, they forward it back to internal host. The difference between NAT firewalls is for how long the NAT tunnel will remain alive, but this does not matter for the audio part of the call, as there is constant flow of audio in both way during a call. This would matter more for the signalling part of the call, which uses the SIP protocol. So I would recommend using TCP for SIP as the TCP session has a default timeout of 900s, making the keep alive messages less frequently needed.
Now some applications you mentioned do not use SIP for session initiation, and hence have proprietary ways of signalling.
Other applications take advantage of something called 'hole punching' to allow client-to-client direct communication (or peer-to-peer) such as Skype. The advantage of these is that the server does not stay in the middle of the voice stream, and this can effectively reduce latency, making TCP a potential choice for the audio stream.
The guys behind development of Asterisk, the famous opensource PBX, have realized the problems in SIP which require having lots of ports open, and they have developed their own protocol, called IAX, to transmit signalling and media over one port. I would encourage you to consider implementing IAX for your client/server, because it ensures that if a client is able to connect (through signalling), then it's able to make calls.