I am studying l2cap specification. Here I found that channel type connection oriented and connection less uses ACL link which is connection less to transfer packet.
Please help me to understand how connection oriented channel with connection less logical link will process data?
If it is silly question please let me know where I can clarify this doubt.
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My goal is to create a parser for TCP packets that are using a custom spec from the Options Price Reporting Authority found here but I have no idea where to start. I've never worked with low-level stuff and I'd appreciate if I got some guidance.
The huge problem is I don't have access to the actual network because it costs a huge sum per month and all I can work off is the specification. I don't even if it's possible. Do you step by step parse each byte and hope for the best? Do you first re-create some example data using the bytes in the spec and then parse it? Isn't that also difficult since (I think) that TCP spread the data to multiple blocks?
That's quite an elaborate data feed. A quick review of the spec shows that it contains enough information to write a program in either nodejs or golang to ingest it.
Getting it to work will be a big job. Your question didn't mention your level of programming skill, or of your network engineering skill. So it's hard to guess how much learning lies ahead of you to get this done.
A few things.
It's a complex enough protocol that you will need to test it with correctly formatted sample data. You need a fairly large collection of sample packets in order to mock your data feed (that is, build a fake data feed for testing purposes). While nothing is impossible, it will be very difficult to build a bug-free program to handle this data without extensive tests.
If you have a developer relationship to the publisher of the data feed, you should ask if they offer sample data for testing.
It is not a TCP / IP data feed. It is an IP multicast datagram feed. In IP multicast feeds you set up a server to listen for the incoming data packets. They use multicast to achieve the very low latencies necessary for predatory algorithmic trading.
You won't use TCP sockets to receive it, you'll use a different programming interface called UDP datagrams
If you're used to TCP's automatic recovery from errors, datagrams will be a challenge. With datagrams you cannot tell if you failed to receive data except by looking at sequence numbers. Most data feeds using IP and multicast have some provision for retransmitting data. Your spec is no exception. You must handle retransmitted data correctly or it will look like you have lots of duplicate data.
Multicast data doesn't move over the public network. You'll need a virtual private network connection to the publisher, or to co-locate your servers in a data center where the feed is available on an internal network.
There's another, operational, spec you'll need to cope with to get this data. It's called the Common IP Multicast Distribution Network Recipient Interface Specification. This spec has a primer on the multicast dealio.
You can do this. When you have made it work, you will have gained some serious skills in network programming and network engineering.
But if you just want this data, you might try to find a reseller of the data that repackages it in an easier-to-consume format. That reseller probably also imposes a delay on the data feed.
I have a Node application that will use RabbitMQ and I am using amqplib to access it. I understand that TCP/IP connections to RabbitMQ and expensive, channels are cheap so create one connection and then multiple channels.
What I am slightly confused about is how I rate that one connection that can be used across the application? Most tutorials seems to indicate that a connection is open for a purpose and then closed again only to be opened again when next required.
I would think that this would result in multiple connections if multiple users were attempting an action that required RabbitMQ access at the same time.
I suppose your users are only connected to your server which is different to each user is connected directly to rabbitmq. Then, server-side, you can keep only one connection to rabbitmq.
For that purpose, I would recommend to create a module to keep track of your connection across your application.
Note that AMQP has a heartbeat feature :
AMQP 0-9-1 offers a heartbeat feature to ensure that the application layer promptly finds out about disrupted connections (and also completely unresponsive peers). Heartbeats also defend against certain network equipment which may terminate "idle" TCP connections.
By default, amqplib's hearbeat is set to 0 meaning no heartbeat.
I'm new for this technology, can somebody help me to know about some doubt?
Q-1. What is the size of CoAP packet?
(I know there is 4 byte fixed header, but what is the maximum size limit including header, option and payload?)
Q-2. Is there any concept for Keep Alive like MQTT?
(It works on UDP for how much time it keeps open the connection, is there any default time or it keeps open every time when we send packet?)
Q-3. Can we use CoAP with TCP?
(Main problem with it CoAP is it works on UDP, is there any concept like MQTT QoS? Let's say a sensor publishes some data every one second, if subscriber goes offline, is there any surety in CoAP that subscriber will get all the data when it come online?)
Q-4. What is the duration of connection?
(CoAP supports publish/subscribe architecture, may be it needs connection open all the time, is it possible with CoAP whether it is based on UDP.)
Q-5. How does it discover the resources?
(I have one gateway and 5 sensors, how will these sensors connect to the gateway? Will the gateway find these sensors? Or will sensors find the gateway?)
Q-5. How does sensor register with gateway?
Please help me, I really need answer. I'm all new for these kind of things and suggest me something for implementation point of view.
Thanks.
It Depends:
Core CoAP messages must be small enough to fit into their link-layer packets (~ 64 KiB for UDP) but, in any case the RFC states that:
it SHOULD fit within a single IP packet to avoid IP fragmentation (MTU of 1280 for IPv6). If nothing is known about the size of the headers, good upper bounds are 1152 bytes for the message size and 1024 bytes for the payload size;
or less to avoid adaptation layer fragmentation (60-80 bytes for 6LoWPAN networks);
if you need to transfer larger payloads, this IETF draft extends core CoAP with new options for transferring multiple blocks of information from a resource representation in multiple request-response pair (so you can transfer more than 64KiB per message).
I never used MQTT, in any case CoAP is connectionless, requests and responses are exchanged asynchronously over UDP or DTLS. I suppose that you are looking for the observe functionality: it enables CoAP clients to "subscribe" to resources and servers to send updates to subscribed clients over a period of time.
There is an IETF draft describing CoAP over TCP, but I don't know how it interacts with the observe functionality: usually It follows a best-effort approach, it just happens that the client is considered no longer interested in the resource and is removed by the server from the list of observers.
The observe stops when the server thinks that the client is no longer interested in the resource or when the client ask to unsubscribe from the resource.
There is a well-known relative URI "/.well-known/core". It is defined as a default entry point for requesting the list of links about resources hosted by a server. Here for more infos.
Look at 5.
I understand many of the fine details of NAT hole punching, ICE, and SIP VOIP calls. I've answered quite a few questions on SO on these topics. Now I have a question.
I am trying to understand the need for the RE-INVITE message that is documented for SIP+ICE after the call is already established.
Assume a topology of VOIP devices that signal over SIP and using ICE (with STUN/TURN) for establishing media connectivity. After the ICE connectivity checks are performed, both endpoints should have ascertained the best address candidate pairings (IP,port) and should be ready to stream media in both directions.
But my experience with SIP and plenty of documentation suggests that after the callee sends a 200 OK message to indicate he's in the answered state, the caller is to expected send a RE-INVITE with an SDP containing the specific address candidate selected by the connectivity checks.
Some links that describe the RE-INVITE with ICE are here and here (step 8). Rosenberg's tutorial (page 30) discusses that the RE-INVITE "ensures that middleboxes have the correct media address". I'm not sure why that's important.
Upon receiving a RE-INVITE, is the callee expected to reconfigure his ICE stack to switch sockets or addresses based on the new SDP received? Or is the RE-INVITE just a protocol formality to formally acknowledge the call has been established? If the RE-INVITE step was skipped and both sides started streaming media, what could go wrong?
The reason why I ask is because I am exploring using ICE over a signaling service that is not SIP. I'm trying to figure out if the RE-INVITE needs to be emulated.
One example of middlebox, which may be interested in the result of the ICE negotiation is a bandwidth manager.
Imagine an enterprise deployment, with endpoints inside the corporate firewall and others roaming around on the Internet, or behind private home firewalls. The deployment also includes a publicly accessible TURN server. Then let's imagine an endpoint inside the corporate firewall making a call. If the destination happens to be reachable on the same network, the call will go host to host and the TURN server will not be used (i.e. bindings will be de-allocated). This is a local network, and no bandwidth limitation needs to be imposed. On the other hand, if the call goes out to a roaming endpoint, then the TURN server will get involved, and data will flow through the corporate firewall, through what probably is a limited bandwidth uplink. We can very well imagine some policy that would want to limit the call's bandwidth. One way of doing it, is for the bandwidth manager to remain in the signaling path (using SIP record-routes) and look into the SDP of any INVITE going through, re-writing bandwidth values depending on media addresses.
I believe the general intention was, that legacy equipment of that type, which is not ICE aware, should keep working in a mixed environment introducing ICE endpoints. To keep this backwards compatibility, ICE should only introduce new processing (STUN checks, etc), but from the point of view of non-ICE-aware elements, it should not remove any existing rules (re-INVITEs are still needed to change the destination of a media stream).
In Rosenberg's tutorial I believe the re-INVITE is sent because the sockets chosen by ICE are different from those in the media and connection (m/c-lines) lines of the original SDP AND in order for any network elements that are in between the two user agents to be informed of the actual sockets that will be used RTP a re-INVITE is sent with the ICE selected socket(s) in the media and connection lines of the SDP.
If the ICE selected sockets were the same as the ones in the original INVITE request's SDP m/c lines then there would be no need for the re-INVITE request. Or if you know there are no "middleboxes" that need to be informed of the changes to the RTP sockets then there would also be no need to send the re-INVITE.
I've been looking into skypes protocol or what people can make out since its a propriety protocol. I've read "An analysis of the skype peer-to-peer internet telephony protocol", though it is old it discusses a certain property which I'm looking to recreate in my own architecture. What I'm interested in is during video a conference, data is sent to one machine (the one most likely with the best bandwidth and processing power) which then redistributes to the other machines.
What is not explained is what happens when the machine receiving and sending the data has unexpectedly dropped out. Of course rather than drop the conference it would be best to find another machine to carry on receiving and distributing the data. Is there any documentation on how this performed on skype or a similar peer-to-peer VoIP?
Basically I'm looking for the fastest method to detect when a "super peer" unexpectedly drops out and quickly migrating operations to another machine.
You need to set a timeout (i.e., limit) and declare that if you don't receive communication within then, the communication is either dead (no path between the peers, reachability issue) or the remote peer is down. There is no other method.
If you have direct tcp or other connection to the super peer, you can catch events telling you the connection dies too. If your communication is relayed, and your framework automatically attempt to find a new route to your target peer, it will either find one or never find out. Hence, the necessity for a timeout.
If none hears about someone for some time, they are finally considered/declared dead.