I need to implement a CAN protocol which is used in Automobile Industry.I wanted to know if it is possible to get the CAN data packets through WiFi using NSNetService/NSNetServiceBrowser class. In Apple's Documentation, everytime they are about to describe about NSNetService/NSNetServiceBrowser
class they say "Bonjour network services" where Bonjour is a protocol implemented by apple itself.
So, I think of course it will detect those i.e Bonjour Data Packets.But now, I am not sure if it will do the same in case of CAN data packets.Hence my question is : Is CAN data packet compatible with NSNetService class so that I can use it directly? If not, is there a way to detect CAN data packets coming from a WiFi network?
Further I got a reply on Apple Developer Forum that if CAN data packets were wrapped up in some High Level Protocol like UDP then NSNetServiceBrowser class MIGHT read it. Now my question is if my CAN data packet was in UDP packet(lets say), is NSNetServiceBrowser class capable of detecting those data packets since NSNetServiceBrowser asks for input on service domain Name and service domain type?
Thanks in Advance.
iParag
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I'm writing a server that, among other things, needs to be constantly sending data in different multicast addresses. The packages being sent might be received by a client side (an app) which will be switching between the mentioned addresses.
I'm using Perfect (https://github.com/PerfectlySoft/Perfect) for writing the server side, however had no luck using the Perfect-Net module nor using CocoaAsyncSocket. How could i implement both the sender and the receiver using swift? Any could snippet would be really useful.
I've been reading about multicasting and when it comes to the receiver, i've notice that in most languages (i.e. java or c#) the receiver often indicates a port number and a multicast ip-address, but when is the connection with the server being made? When does the socket bind to the real server ip-address?
Thanks in advance
If we talk about the TCP/IP stack, only IP and UDP support broadcasts and multicasts. They're both connectionless, and this is why you see only sending and receiving to special multicast addresses, but no binds and connects. You see it in different languages because (a) protocols are language-agnostic and (b) most implementations put reasonable efforts in trying to be compatible with BSD sockets interface.
If you want that true multicast, you'll need to find a swift implementation of sockets that allow setting options. Usual names for this operation is setsockopt. Multicast sender side doesn't need anything beyond a basic UDP socket (I suggest using UDP, not IP), while sender needs to be added to a multicast group. This Python example pretty much describes it.
However, it's worth noting that routers don't route broadcasts and multicasts. Hence you cannot use it over internet. If you need to use internet in your project, I'd advise you to use TCP - or websockets if your clients are browsers - and send messages to "groups" of them manually.
I guess you actually want Perfect-Kafka or Perfect-Mosquitto - Message Queue which allows a server to publish live streams to the client side subscribers. Low-level sockets will not easily fulfill your requirement.
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I always hear about the layers of internet and i vaguely understand this. But, what confuse me most is that transport layer (including TCP protocol) lies on top of the internet layer(including IP protocol)..
What does this mean? For one who has a foggy understanding of the internet mechanism (I'm not a CS student or something I am just a hobby programmer)
The picture I have about the internet is that the network card sends/receives signals (packets) from the internet through wired connection / wifi then the OS using socket API sends/receives these packets acting as a layer between the hardware and the application which in turn uses some high-level protocol such as HTTP to interpret the data transferred - these protocol usually provided by languages e.g. python or java
.. I guess then that IP and TCP protocol are used at the level of the socket API? but I need more details ? I hope the explanation be in terms of coding/programming/implementation because abstractions used in this area confuse me.
Thank you , and sorry for my bad English
This is part of a layered solution to solve networking. Each layer has its own functionality:
IP (Internet Protocol) is in charge of delivering a packet (or datagram) from one interface, in one machine, with an IP address assigned to another interface in the same or other machine (node). Both nodes can be in the same LAN or different LAN connected through different paths (LAN's and routers). Basically it will make the packet get from source IP to destination IP. It provides a best-effort services, it doesn't assure the IP packet is going to arrive, it can be lost in the middle.
Above layer 3 or IP in the so-called TCP/IP stack, there is the transport layer. Its main functionality is to multiplex the lower layer (IP) service (take a packet from src to dst) among different applications. This is why in all transport layer protocols there is the concept of port or more generically Transport Service Access Point (TSAP). UDP, TCP, SCTP do that. UDP provides an unreliable service to the application. TCP provides a connected, reliable transport service to the application. This layer will make a message sent from application A in node Y reach application A1 in node Z, either reliably or unreliably (while IP only takes care of carrying the packet from node Y to node Z).
You will need to read a little about the OSI layered model and the TCP/IP layered model.
If you need to get more info I can address you to a training I have about IPv6 with a good introduction to networking: http://www.slideshare.net/rodolk/networking-tcpip-stack-introduction-ipv6
TCP is a protocol, known as "Transmission Control Protocol" - by specification it has features in place which makes sure that transmitted data is checked. On the other hand, there are things such as UDP, aka "User Datagram Protocol" which also works on top of IP - by specification this method does not check any transmitted data, so it's less useful where files must be fully intact (more utilised for video streaming, where some lost frames is acceptable, as opposed to binary file transfers where incorrect data means corruption and the whole file would be useless).
On to IP, IP is an addressing protocol, allowing a network to address and communicate with any machine that lives within it. IP stands for Internet Protocol, and it defines the fundamental way that two machines communicate over the "internet". It does not define how communications are handled, in ways such as being checked for data integrity, etc.
So, to summarise, the TCP and UDP are just extensions of IP. It is entirely possible, however, to have a socket based TCP or UDP connection, and I expect it's also possible to have some sort of MAC address protocol (as opposed to an IP address protocol). I don't know of any protocols which are similar to IP, but I imagine they do exist. In reality, using TCP over something other than IP is entirely unlikely. If you're going to the effort to create a custom protocol, chances are you'll want it fully custom and won't want to stick to design specifications designed for another protocol layer.
Note that calling it a "TCP/IP" connection is probably only ever used for legacy reasons. A lot of terms like this still exist because before the technology "bubble" growth, there were competing alternatives to IP. Even today, there is IPv6 which is technically an alternative to IPv4. It's also possible that we might one day outgrow IPv6, and at that point in time, there could be something other than IP to worry about.
IS there a way I can in user-space get notification about a packet being dropped at Layer-2 in 802.11.
According to my understanding what happens is, when a packet is sent out on the medium, there are Layer-2 ACKs which are received if it is delivered correctly (if not,it does the retransmission and ultimately drops the packet if not delivered after several retries..)
I want to be able to access this notification (in user-space)and change the behavior of packet transmission.
I want to be able to send the packet to another host from the FIB rather than dropping the packet.
I have read about libpcap libraries and netfilter hooks which allows me to capture packet and inject them back on the networking stack..
But I'm not able to find hooks (if any, for the wireless stack) to help me capture the packet notification in Layer-2.
Please correct me if I'm not understanding something correctly. Also, any heads-up or links to read would be great.
No, you cannot, at least not using the standardised sockets interfaces. 802.11 is a link layer, and the sockets API is strictly link-layer agnostic: unless it's going to work on all link layers, it's not in sockets. There are good reasons for that: the kind of cross-layer interaction that you envision has been tried many times, and it's always turned out more trouble than it's worth.
You didn't give us any details about the application — but the best solution is most probably to change your application-layer protocol to send explicit acknowledgments, and send your data over the fallback route when you fail to receive an ACK.
What is the purpose for using sip with VOIP ? is it just to know the remote ip address ?
If i know (by server) the remote IP address ,then established a direct TCP socket connection for call negotiation, and send the media over RTP protocol ,so am i still need sip protocol ? or how can sip help me here?
The Session Initiation Protocol does rather a lot more than find out a remote IP address/port/transport triple.
It lets two parties
negotiate the media streams (including codecs and transports) and
establish commonly understood extensions to the protocol.
It also describes how to build scalable infrastructure (proxies, using SRV and NAPTR records, back to back user agents), location services and a host of other details that go into making a voice (or any other kind of) call to arbitrary third parties.
then you have implemented a sip alternative. sip (session instanciation protocol) does just control the "phone call". if you want to do that on your own, why not? the only problem would be that there are many sip clients and just one (or few) clients using your protocol.
SIP allocates a IP:port to a voip call. The RTP flows (one for each direction) will then use this IP:port as a destination address. If you have only one static RTP flow to send to your server, it may be useful and ok to do what you said.
Otherwise, if there are many clients, or if your system has to change a lot, it's better to use a polished protocol which will dynamically allocate ports and establish your sessions.
nobody forces you to implement a standard.
e.g. why do you implement the media stream in RTP? most likely because you already have code that "talks" RTP (e.g. a library, or a raedymade application).
the nice thing about standards is, that it will work "out-of-the-box" with all other applications implementing the same standard.
if it is an open standard, there's another nice thing: other people have already spent a lot of brain power into getting the implementation right. you don't need to fall into the same problems.
Does anyone know why should I use Winpcap and not just .Net sockets to sniff packets on my local pc?
TY
Sockets (.NET, Winsock, etc.) normally collect at layer 7, the Application layer. That is, whatever is sent by the sender is what is received by the receiver. All of the various headers that are added automatically on the sending side are stripped off by the time the receiver reads the data from the socket.
It is possible to configure a socket to be a raw socket, in which case, you can see all of the headers down to layer 3, the Network layer. Further still, you can put the raw socket in promiscuous mode, which allows you to see all of the traffic on the network, not just the packets destined for your machine. But even this is limited. For example, when you configure the raw socket, you specify the protocol type to use, e.g., IP, ICMP, etc. This limits the socket to "seeing" packets that adhere to that protocol. I have been unable to figure out how to make the socket see all packets at layer 3 regardless of protocol.
Winpcap operates as a device driver at layer 2, the Data Link layer. In this case, you see literally all of the packets on the network with full headers down to layer 2. Winpcap also offers filtering capability so you can narrow down the packets that are reported to you based on whatever criteria you provide.
As far as choosing between them, it really boils down to the requirements of your specific task. If you are trying to implement any kind of realistic network analysis capability, you'll be hardpressed to do that with just sockets. Winpcap makes more sense in that case. However, if you are only interested in IP packets, for example, then sockets will work fine for that.
As far as I understanf .Net sockets are an IPC to communicate between 2 processes. While winpcap is a library that help you to access the data link layer an sniff pacquets going through your network hardware (or virtual) devices on your machine. Data link layer allow to get the data on any socket (.Net or not) created on your system.