I'm using a TK102 GPS localizer. Along with it, I got only simple end-user docs. No API, dev specs or similar for writing code that will use this localizer.
I was told that it uses UDP. So I wrote a simple PHP listener. But either localizer is not using UDP or something is wrong in communication between it and server. Listener works fine (gets UDP packets from other clients) and localizer is sending something (I'm being charge by GSM operator for GPRS transmission), but the data it sends, doesn't reach server.
I asked about server or networking issues on Unix/Linux and SuperUser. Here I would only ask, if someone knows any API/dev-specs for this localizer, so I can check, if it really uses UDP or if I haven't made any other error (in configuration for example).
The localizer and its clones
We're talking about Xexun TK102 Tracker here. The original one, because there are many clones under other companies from China, selling similar GPS localizer, with the same cover and logo, but with:
less performance electronics on-board (for example -- able to report location once per 20 or 30 seconds, not once per 5 seconds like in original one),
the ones that are sending lesser information (lack of direction/bearing, altitude, number of satelites used for location fix and many more),
units using different format of data or non-standard transmission protocol for sending it (for example, cheaper units are unable to use UDP protocol and are transmiting data through TCP protocol, using packets that not always follows standards or definictions.
Coban and Kintech are only two of many clones sold on eBay and in e-shops, claiming to be original Xexun trackers.
On the other hand, original Xexun and some clones (like Coban for example) are harder to control from own script, because they require a correct answer from the server, where data is sent over GPRS. If unit does not receive such reply, it breaks connection. The cheapes unit does not have this checking and will always sent location data to specified IP address over provided port.
Product description
Here is product description of original Xexun localizer (and here is a clone under Kintech name).
Possible buyer must be very careful (and should secure return policy, for which buying directly in China is not recommended) as there are many reports about sellers claiming to sell original Xexun device and sending a clone actually.
Though this device is five years old, it is still sold at many places (including eBay), but even at theses sources it is very hard to get anything worth for developers, except some simple, very basic user guide.
I have confirmed information (from two different sources) that there is no official API available for this device. The only option is to Google around, ask other users or use forums (see below).
If you own original Xexun localizer, you may try to contact company international departament and ask their technicians to include some changes to device source code and to send you updated firmware, with your changes - wow! That was confirmed by company itself.
Forum
I found a perfect forum for TK102 device, with a lot of questions and answers:
here is a general forum on TK102 device (kept alive for 4,5 year with 171 pages and 2000+ posts!),
here you'll find more specific topic on receiving data from this localizer,
this forum is also about TK102 unit, but it is entirely in French.
There are many other devices dissussed and in general, this is the biggest forum in the world, with topics for localizers and simillar information.
GPRS Protocol Specs
In general, any TK102 related devices is opening a socket for a direct TCP transmission (original one can be switched to use UDP protocol). Data is being transsmited over port specified by user, in configuration and using GPRS only (requires SIM card with enabled GPRS, there is no way to use WiFi).
Sending frequency, format and amount of data being send, entirely depends on kind of device is being used -- it is more extensive and more configurable in original one than in clones.
Using FileDropper I shared GPRS Protocol Specification for TK102 Geolocalizer. It contains basic information on how to setup TK102 (and possible all its clones) to send location over GPRS. And what sort of data you should except to receive from in, on server side. This could be useful for someone.
BTW: If links goes dead, contact me for a reupload or sending it over e-mail
Correct server response problem
Make sure, if you're using correct data transmission protocol! Many (really many) cheap clones uses TCP, while only original TK102 allows switching to UDP. This is convenient, because you need really basic server configuration to handle TCP connections, while you have to use specific server-side software (like node.js) or specific configuration (open to certain ports) to handle UDP. But the key thing is to determine correct protocol, as listening to TCP data, while your localizer sends UDP, will most certainly fail.
Take into consideration, that many TK102 clones requires a correct response from the server after each data, it send. It breaks connection after sending some welcome garbage UDP packet, as it does not receive response, it waits for.
It is quite hard (quite impossible?) to find any guide to many of these clones, on what kind of responses server should sent. This often leads into situation of developer being unable to estabilish two-way communication between server and localizer. Many localizers are sold to be used only via SMS communication or throughs paid services that had signed and agreement with producer and received protocol specification that contains valid responses server should generate for particular TK102 clone.
Double check, if this is not source of problem, if you can't communiacte with your localizer from your app.
You can check some models protocol specs here:
http://www.traccar.org/docs/protocol.jsp
Related
I was reading TURN server RFCs. All related RFCs ( 5766 and the more recent 8656) support Channel mechanism to avoid the 36 bytes overheads of STUN headers (Section 2.5 of RFC 5766) required for the send/data approach:
For some applications (e.g., Voice over IP), the 36 bytes of overhead
that a Send indication or Data indication adds to the application
data can substantially increase the bandwidth required between the
client and the server. To remedy this, TURN offers a second way for
the client and server to associate data with a specific peer.
For WebRTC, clearly there is no point in using the send/data mechanism. How do browsers choose between the two mechanisms for relaying? Is send/data a fallback? Will support for Channels alone in a TURN server be sufficient for WebRTC use-case?
They will usually do SendIndications while waiting for the Channel to be created.
SendIndications also are important if you get something on the relay before the Channel is created. Some clients only create the Channel when they send and not right when the permission is created.
Firefox doesn't support TURN channels: https://bugzilla.mozilla.org/show_bug.cgi?id=857736
Chrome also uses send/binding indications until ICE is done (presumably to avoid the overhead of creating channels which are not used later)
Don't rely on partial implementations of a spec, that won't work.
I am working with pcap in an OS X application to understand packet analysis.
I am working with a app https://github.com/jpiccari/MacAlyzer
but I am getting only raw data but I want to differentiate every domain request into separate and clear way to read request and response value. Please guide me the way to how to develop an application with pcap.
I have tried some code but they translate data into hex format. How do I convert that data into meaningful request and response objects like Charles and Fiddler show?
MacAlyzer wasn't developed for your needs. I know because I'm the author. As already stated, Charles and Fiddler are web proxies and work entirely different (and serve different purposes).
Diving a bit deeper into your question, communication between client and server happens IP-to-IP and not domain-to-domain. Domain information is not contained in the packets at the either the IP or TCP level. Instead computers request domain-to-IP lookup information which is then stored and communication is carried out using the client and server IP addresses.
MacAlyzer, and really libpcap, don't have sophisticated packet dissection (like say Wireshark) and cannot display packet information as verbosely as other programs. Before I lost interest in the project I was planning a library that would allow much richer packet dissection and analysis, but free time became very limited.
As for adding domain information to MacAlyzer, I'll explain at a high-level since it seems you know what you're doing. To include domain information instead of IP address in the Source and Destination columns you could edit function ip_host_string() in ip.m. This function controls how the client and server addresses are displayed. Modifying it to lookup the hostname from IP address and returning the resulting string would cause the domains to be displayed instead of IP addresses.
If you come up with some nice updates, consider submitting a pull request.
Here is the food for thoughts:
http://www.binarytides.com/packet-sniffer-code-c-linux/
Anyway, you will need to use C. Therefore, check the codes of the includes, for example:
http://www.eg.bucknell.edu/~cs363/2014-spring/code/tcp.h
Here is the documentation of "pcap":
http://www-01.ibm.com/support/knowledgecenter/#!/ssw_aix_71/com.ibm.aix.basetrf1/pcap_close.htm
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.
I'm looking to write a toy application for my own personal use (and possibly to share with friends) for peer-to-peer shared status on a local network. For instance, let's say I wanted to implement it for the name of the current building you're in (let's pretend the network topology is weird, and multiple buildings occupy the same LAN). The idea is if you run the application, you can set what building you're in, and you can see the buildings of every other user running the application on the local network.
The question is, what's the best transport/network layer technology to use to implement this?
My initial inclination was to use UDP Multicast, but the more research I do about it, the more I'm scared off by it: while the technology is great and seems easy to use, if the application is not tailored for a particular site deployment, it also seems most likely to get you a visit from an angry network admin.
I'm wondering, therefore, since this is a relatively low bandwidth application — probably max one update every 4–5 minutes or so from each client, with likely no more than 25–50 clients — whether it might be "cheaper" in many ways to use another strategy:
Multicast: find a way to pick a well-known multicast address from 239.255/16 and have interested applications join the group when they start up.
Broadcast: send out a single UDP Broadcast message every time someone's status changes (and one "refresh" broadcast when the app launches, after which every client replies directly to the requesting user with their current status).
Unicast: send a UDP Broadcast at application start to announce interest, and when a client's status changes, it sends a UDP packet directly to every client who has announced. This results in the highest traffic, but might be less likely to annoy other systems with needless broadcast packets. It also introduces potential complications when apps crash (in terms of generating unnecessary traffic).
Multicast is most certainly the best technology for the job, but I'm wondering if the associated hassles are worth avoiding since this is just a "toy application," not a business-critical service intended for professional network admin deployment and configuration.
We are running a course in robotics and Xbee is the most favorite communication protocol for the student. In last two years we helped them build around 62 various projects (40 more in pipeline).
All most all the projects involve sending different kind of data to the bot. Sometimes it is a 1 byte command where as sometimes it is a long string to be interpreted. Sometimes we face the issue of addressing a bot when one xbee is used in broadcast mode to send messages to a particular bot among several. Students use their creativity to address this issue each time.
I personally feel this is reinvesting the wheel. I wonder if any higher level protocol proposals exist for serial port communication and if there isn't any specific protocol design I wonder if if the worth designing one for the student needs.
Do you mean internal only protocol of your system? If yes, often embedded software engineers incline to roll their own protocols. Most of them talks that it lets them make most optimal system.
It is not ideal approach. I agree with you that it's good for students to learn good examples.
Unfortunately I don't know any protocol stack fitting well robotics application. But I advice you to try google's protocol buffer system, its able to simplify most efforts of building protocols engines, and it works with plain c too.
You can implement Modbus ASCII if you want to go with a standard protocol that's already open.
Comli is a master/slave protocol that is used in some older devices or when it is not possible to use ethernet. You can probably get the specification from ABB if you ask - it's no secret.
That said you can put an OPC server/client architecture on top of that to get a bit more powerful communication e.g.
+--------------+ +--------------+ +--------+
| OPC UA Client| -- | OPC UA Server| -comli- | Device |
+--------------+ +--------------+ +--------+
This would make your OPC UA client protocol indepedent which makes things a bit easier down the road.
Modbus is another serial protocol that is used a lot
I believe OPC will give you the highlevel operation that you want.
see
www.opcfoundation.org
www.abb.com
PS. OPC UA is not the same as the old OLE version and thus has nothing to do with COM/DCOM
Like mjh2007 said, Modbus is standard, open and easy. The only problem I can see is if you want the robot to respond "quickly" to a command, since serial Modbus uses timeouts to detect the end of a packet. You can get around this by ignoring the timeout requirements and calculating the expected size of a packet based on it's function code and parameters as you are receiving it, then you can start processing the command immediately upon receiving the last byte and verifying any checksums. This page has some more details on implementing such a scheme.
Be sure to make use of the XBee module's "Transmit Explicit" frame (type 0x11) running in API mode with ATAO set to 1. You can unicast to a particular bot on your network, instead of always broadcasting frames. On a mesh ZigBee network, you want to avoid broadcasts as much as possible.
I'm guessing you're either using "AT mode" for sending raw data, or using "API mode" with ATAO set to 0 (sometimes referred to as "transparent serial").
If you look at that frame type (0x11), you'll see that the recipient gets an 0x91 frame that contains multiple fields already (source/destination endpoint, cluster, profile ID). You can re-purpose those fields since you're not trying to do ZigBee networking.