73 skills found · Page 1 of 3
holo-routing / HoloHolo is a suite of routing protocols designed to support high-scale and automation-driven networks.
tildeio / Route RecognizerA lightweight JavaScript library that matches paths against registered routes. It includes support for dynamic and star segments and nested handlers.
rtbrick / BngblasterThe Routing Protocol and BNG Blaster is an open-source network tester for routing and access protocols.
CiscoDevNet / Pathman SrPathman SR is an open-source app to compute paths and deploy routes in Segment Routing-enabled network.
mc36 / FreeRtrfreeRouter - networking swiss army knife
rudderlabs / Rudder TransformerOpen-source, warehouse-first Customer Data Pipeline and Segment-alternative. Collects and routes clickstream data and builds your customer data lake on your data warehouse.
nttcom / PolaStateful PCE Implementation and PCEP Library in Go
SOYJUN / FTP Implement Based On UDPThe aim of this assignment is to have you do UDP socket client / server programming with a focus on two broad aspects : Setting up the exchange between the client and server in a secure way despite the lack of a formal connection (as in TCP) between the two, so that ‘outsider’ UDP datagrams (broadcast, multicast, unicast - fortuitously or maliciously) cannot intrude on the communication. Introducing application-layer protocol data-transmission reliability, flow control and congestion control in the client and server using TCP-like ARQ sliding window mechanisms. The second item above is much more of a challenge to implement than the first, though neither is particularly trivial. But they are not tightly interdependent; each can be worked on separately at first and then integrated together at a later stage. Apart from the material in Chapters 8, 14 & 22 (especially Sections 22.5 - 22.7), and the experience you gained from the preceding assignment, you will also need to refer to the following : ioctl function (Chapter 17). get_ifi_info function (Section 17.6, Chapter 17). This function will be used by the server code to discover its node’s network interfaces so that it can bind all its interface IP addresses (see Section 22.6). ‘Race’ conditions (Section 20.5, Chapter 20) You also need a thorough understanding of how the TCP protocol implements reliable data transfer, flow control and congestion control. Chapters 17- 24 of TCP/IP Illustrated, Volume 1 by W. Richard Stevens gives a good overview of TCP. Though somewhat dated for some things (it was published in 1994), it remains, overall, a good basic reference. Overview This assignment asks you to implement a primitive file transfer protocol for Unix platforms, based on UDP, and with TCP-like reliability added to the transfer operation using timeouts and sliding-window mechanisms, and implementing flow and congestion control. The server is a concurrent server which can handle multiple clients simultaneously. A client gives the server the name of a file. The server forks off a child which reads directly from the file and transfers the contents over to the client using UDP datagrams. The client prints out the file contents as they come in, in order, with nothing missing and with no duplication of content, directly on to stdout (via the receiver sliding window, of course, but with no other intermediate buffering). The file to be transferred can be of arbitrary length, but its contents are always straightforward ascii text. As an aside let me mention that assuming the file contents ascii is not as restrictive as it sounds. We can always pretend, for example, that binary files are base64 encoded (“ASCII armor”). A real file transfer protocol would, of course, have to worry about transferring files between heterogeneous platforms with different file structure conventions and semantics. The sender would first have to transform the file into a platform-independent, protocol-defined, format (using, say, ASN.1, or some such standard), and the receiver would have to transform the received file into its platform’s native file format. This kind of thing can be fairly time consuming, and is certainly very tedious, to implement, with little educational value - it is not part of this assignment. Arguments for the server You should provide the server with an input file server.in from which it reads the following information, in the order shown, one item per line : Well-known port number for server. Maximum sending sliding-window size (in datagram units). You will not be handing in your server.in file. We shall create our own when we come to test your code. So it is important that you stick strictly to the file name and content conventions specified above. The same applies to the client.in input file below. Arguments for the client The client is to be provided with an input file client.in from which it reads the following information, in the order shown, one item per line : IP address of server (not the hostname). Well-known port number of server. filename to be transferred. Receiving sliding-window size (in datagram units). Random generator seed value. Probability p of datagram loss. This should be a real number in the range [ 0.0 , 1.0 ] (value 0.0 means no loss occurs; value 1.0 means all datagrams all lost). The mean µ, in milliseconds, for an exponential distribution controlling the rate at which the client reads received datagram payloads from its receive buffer. Operation Server starts up and reads its arguments from file server.in. As we shall see, when a client communicates with the server, the server will want to know what IP address that client is using to identify the server (i.e. , the destination IP address in the incoming datagram). Normally, this can be done relatively straightforwardly using the IP_RECVDESTADDR socket option, and picking up the information using the ancillary data (‘control information’) capability of the recvmsg function. Unfortunately, Solaris 2.10 does not support the IP_RECVDESTADDR option (nor, incidentally, does it support the msg_flags option in msghdr - see p.390). This considerably complicates things. In the absence of IP_RECVDESTADDR, what the server has to do as part of its initialization phase is to bind each IP address it has (and, simultaneously, its well-known port number, which it has read in from server.in) to a separate UDP socket. The code in Section 22.6, which uses the get_ifi_info function, shows you how to do that. However, there are important differences between that code and the version you want to implement. The code of Section 22.6 binds the IP addresses and forks off a child for each address that is bound to. We do not want to do that. Instead you should have an array of socket descriptors. For each IP address, create a new socket and bind the address (and well-known port number) to the socket without forking off child processes. Creating child processes comes later, when clients arrive. The code of Section 22.6 also attempts to bind broadcast addresses. We do not want to do this. It binds a wildcard IP address, which we certainly do not want to do either. We should bind strictly only unicast addresses (including the loopback address). The get_ifi_info function (which the code in Section 22.6 uses) has to be modified so that it also gets the network masks for the IP addresses of the node, and adds these to the information stored in the linked list of ifi_info structures (see Figure 17.5, p.471) it produces. As you go binding each IP address to a distinct socket, it will be useful for later processing to build your own array of structures, where a structure element records the following information for each socket : sockfd IP address bound to the socket network mask for the IP address subnet address (obtained by doing a bit-wise and between the IP address and its network mask) Report, in a ReadMe file which you hand in with your code, on the modifications you had to introduce to ensure that only unicast addresses are bound, and on your implementation of the array of structures described above. You should print out on stdout, with an appropriate message and appropriately formatted in dotted decimal notation, the IP address, network mask, and subnet address for each socket in your array of structures (you do not need to print the sockfd). The server now uses select to monitor the sockets it has created for incoming datagrams. When it returns from select, it must use recvfrom or recvmsg to read the incoming datagram (see 6. below). When a client starts, it first reads its arguments from the file client.in. The client checks if the server host is ‘local’ to its (extended) Ethernet. If so, all its communication to the server is to occur as MSG_DONTROUTE (or SO_DONTROUTE socket option). It determines if the server host is ‘local’ as follows. The first thing the client should do is to use the modified get_ifi_info function to obtain all of its IP addresses and associated network masks. Print out on stdout, in dotted decimal notation and with an appropriate message, the IP addresses and network masks obtained. In the following, IPserver designates the IP address the client will use to identify the server, and IPclient designates the IP address the client will choose to identify itself. The client checks whether the server is on the same host. If so, it should use the loopback address 127.0.0.1 for the server (i.e. , IPserver = 127.0.0.1). IPclient should also be set to the loopback address. Otherwise it proceeds as follows: IPserver is set to the IP address for the server in the client.in file. Given IPserver and the (unicast) IP addresses and network masks for the client returned by get_ifi_info in the linked list of ifi_info structures, you should be able to figure out if the server node is ‘local’ or not. This will be discussed in class; but let me just remind you here that you should use ‘longest prefix matching’ where applicable. If there are multiple client addresses, and the server host is ‘local’, the client chooses an IP address for itself, IPclient, which matches up as ‘local’ according to your examination above. If the server host is not ‘local’, then IPclient can be chosen arbitrarily. Print out on stdout the results of your examination, as to whether the server host is ‘local’ or not, as well as the IPclient and IPserver addresses selected. Note that this manner of determining whether the server is local or not is somewhat clumsy and ‘over-engineered’, and, as such, should be viewed more in the nature of a pedagogical exercise. Ideally, we would like to look up the server IP address(es) in the routing table (see Section 18.3). This requires that a routing socket be created, for which we need superuser privilege. Alternatively, we might want to dump out the routing table, using the sysctl function for example (see Section 18.4), and examine it directly. Unfortunately, Solaris 2.10 does not support sysctl. Furthermore, note that there is a slight problem with the address 130.245.1.123/24 assigned to compserv3 (see rightmost column of file hosts, and note that this particular compserv3 address “overlaps” with the 130.245.1.x/28 addresses in that same column assigned to compserv1, compserv2 & comserv4). In particular, if the client is running on compserv3 and the server on any of the other three compservs, and if that server node is also being identified to the client by its /28 (rather than its /24) address, then the client will get a “false positive” when it tests as to whether the server node is local or not. In other words, the client will deem the server node to be local, whereas in fact it should not be considered local. Because of this, it is perhaps best simply not to use compserv3 to run the client (but it is o.k. to use it to run the server). Finally, using MSG_DONTROUTE where possible would seem to gain us efficiency, in as much as the kernel does not need to consult the routing table for every datagram sent. But, in fact, that is not so. Recall that one effect of connect with UDP sockets is that routing information is obtained by the kernel at the time the connect is issued. That information is cached and used for subsequent sends from the connected socket (see p.255). The client now creates a UDP socket and calls bind on IPclient, with 0 as the port number. This will cause the kernel to bind an ephemeral port to the socket. After the bind, use the getsockname function (Section 4.10) to obtain IPclient and the ephemeral port number that has been assigned to the socket, and print that information out on stdout, with an appropriate message and appropriately formatted. The client connects its socket to IPserver and the well-known port number of the server. After the connect, use the getpeername function (Section 4.10) to obtain IPserver and the well-known port number of the server, and print that information out on stdout, with an appropriate message and appropriately formatted. The client sends a datagram to the server giving the filename for the transfer. This send needs to be backed up by a timeout in case the datagram is lost. Note that the incoming datagram from the client will be delivered to the server at the socket to which the destination IP address that the datagram is carrying has been bound. Thus, the server can obtain that address (it is, of course, IPserver) and thereby achieve what IP_RECVDESTADDR would have given us had it been available. Furthermore, the server process can obtain the IP address (this will, of course, be IPclient) and ephemeral port number of the client through the recvfrom or recvmsg functions. The server forks off a child process to handle the client. The server parent process goes back to the select to listen for new clients. Hereafter, and unless otherwise stated, whenever we refer to the ‘server’, we mean the server child process handling the client’s file transfer, not the server parent process. Typically, the first thing the server child would be expected to do is to close all sockets it ‘inherits’ from its parent. However, this is not the case with us. The server child does indeed close the sockets it inherited, but not the socket on which the client request arrived. It leaves that socket open for now. Call this socket the ‘listening’ socket. The server (child) then checks if the client host is local to its (extended) Ethernet. If so, all its communication to the client is to occur as MSG_DONTROUTE (or SO_DONTROUTE socket option). If IPserver (obtained in 5. above) is the loopback address, then we are done. Otherwise, the server has to proceed with the following step. Use the array of structures you built in 1. above, together with the addresses IPserver and IPclient to determine if the client is ‘local’. Print out on stdout the results of your examination, as to whether the client host is ‘local’ or not. The server (child) creates a UDP socket to handle file transfer to the client. Call this socket the ‘connection’ socket. It binds the socket to IPserver, with port number 0 so that its kernel assigns an ephemeral port. After the bind, use the getsockname function (Section 4.10) to obtain IPserver and the ephemeral port number that has been assigned to the socket, and print that information out on stdout, with an appropriate message and appropriately formatted. The server then connects this ‘connection’ socket to the client’s IPclient and ephemeral port number. The server now sends the client a datagram, in which it passes it the ephemeral port number of its ‘connection’ socket as the data payload of the datagram. This datagram is sent using the ‘listening’ socket inherited from its parent, otherwise the client (whose socket is connected to the server’s ‘listening’ socket at the latter’s well-known port number) will reject it. This datagram must be backed up by the ARQ mechanism, and retransmitted in the event of loss. Note that if this datagram is indeed lost, the client might well time out and retransmit its original request message (the one carrying the file name). In this event, you must somehow ensure that the parent server does not mistake this retransmitted request for a new client coming in, and spawn off yet another child to handle it. How do you do that? It is potentially more involved than it might seem. I will be discussing this in class, as well as ‘race’ conditions that could potentially arise, depending on how you code the mechanisms I present. When the client receives the datagram carrying the ephemeral port number of the server’s ‘connection’ socket, it reconnects its socket to the server’s ‘connection’ socket, using IPserver and the ephemeral port number received in the datagram (see p.254). It now uses this reconnected socket to send the server an acknowledgment. Note that this implies that, in the event of the server timing out, it should retransmit two copies of its ‘ephemeral port number’ message, one on its ‘listening’ socket and the other on its ‘connection’ socket (why?). When the server receives the acknowledgment, it closes the ‘listening’ socket it inherited from its parent. The server can now commence the file transfer through its ‘connection’ socket. The net effect of all these binds and connects at server and client is that no ‘outsider’ UDP datagram (broadcast, multicast, unicast - fortuitously or maliciously) can now intrude on the communication between server and client. Starting with the first datagram sent out, the client behaves as follows. Whenever a datagram arrives, or an ACK is about to be sent out (or, indeed, the initial datagram to the server giving the filename for the transfer), the client uses some random number generator function random() (initialized by the client.in argument value seed) to decide with probability p (another client.in argument value) if the datagram or ACK should be discarded by way of simulating transmission loss across the network. (I will briefly discuss in class how you do this.) Adding reliability to UDP The mechanisms you are to implement are based on TCP Reno. These include : Reliable data transmission using ARQ sliding-windows, with Fast Retransmit. Flow control via receiver window advertisements. Congestion control that implements : SlowStart Congestion Avoidance (‘Additive-Increase/Multiplicative Decrease’ – AIMD) Fast Recovery (but without the window-inflation aspect of Fast Recovery) Only some, and by no means all, of the details for these are covered below. The rest will be presented in class, especially those concerning flow control and TCP Reno’s congestion control mechanisms in general : Slow Start, Congestion Avoidance, Fast Retransmit and Fast Recovery. Implement a timeout mechanism on the sender (server) side. This is available to you from Stevens, Section 22.5 . Note, however, that you will need to modify the basic driving mechanism of Figure 22.7 appropriately since the situation at the sender side is not a repetitive cycle of send-receive, but rather a straightforward progression of send-send-send-send- . . . . . . . . . . . Also, modify the RTT and RTO mechanisms of Section 22.5 as specified below. I will be discussing the details of these modifications and the reasons for them in class. Modify function rtt_stop (Fig. 22.13) so that it uses integer arithmetic rather than floating point. This will entail your also having to modify some of the variable and function parameter declarations throughout Section 22.5 from float to int, as appropriate. In the unprrt.h header file (Fig. 22.10) set : RTT_RXTMIN to 1000 msec. (1 sec. instead of the current value 3 sec.) RTT_RXTMAX to 3000 msec. (3 sec. instead of the current value 60 sec.) RTT_MAXNREXMT to 12 (instead of the current value 3) In function rtt_timeout (Fig. 22.14), after doubling the RTO in line 86, pass its value through the function rtt_minmax of Fig. 22.11 (somewhat along the lines of what is done in line 77 of rtt_stop, Fig. 22.13). Finally, note that with the modification to integer calculation of the smoothed RTT and its variation, and given the small RTT values you will experience on the cs / sbpub network, these calculations should probably now be done on a millisecond or even microsecond scale (rather than in seconds, as is the case with Stevens’ code). Otherwise, small measured RTTs could show up as 0 on a scale of seconds, yielding a negative result when we subtract the smoothed RTT from the measured RTT (line 72 of rtt_stop, Fig. 22.13). Report the details of your modifications to the code of Section 22.5 in the ReadMe file which you hand in with your code. We need to have a sender sliding window mechanism for the retransmission of lost datagrams; and a receiver sliding window in order to ensure correct sequencing of received file contents, and some measure of flow control. You should implement something based on TCP Reno’s mechanisms, with cumulative acknowledgments, receiver window advertisements, and a congestion control mechanism I will explain in detail in class. For a reference on TCP’s mechanisms generally, see W. Richard Stevens, TCP/IP Illustrated, Volume 1 , especially Sections 20.2 - 20.4 of Chapter 20 , and Sections 21.1 - 21.8 of Chapter 21 . Bear in mind that our sequence numbers should count datagrams, not bytes as in TCP. Remember that the sender and receiver window sizes have to be set according to the argument values in client.in and server.in, respectively. Whenever the sender window becomes full and so ‘locks’, the server should print out a message to that effect on stdout. Similarly, whenever the receiver window ‘locks’, the client should print out a message on stdout. Be aware of the potential for deadlock when the receiver window ‘locks’. This situation is handled by having the receiver process send a duplicate ACK which acts as a window update when its window opens again (see Figure 20.3 and the discussion about it in TCP/IP Illustrated). However, this is not enough, because ACKs are not backed up by a timeout mechanism in the event they are lost. So we will also need to implement a persist timer driving window probes in the sender process (see Sections 22.1 & 22.2 in Chapter 22 of TCP/IP Illustrated). Note that you do not have to worry about the Silly Window Syndrome discussed in Section 22.3 of TCP/IP Illustrated since the receiver process consumes ‘full sized’ 512-byte messages from the receiver buffer (see 3. below). Report on the details of the ARQ mechanism you implemented in the ReadMe file you hand in. Indeed, you should report on all the TCP mechanisms you implemented in the ReadMe file, both the ones discussed here, and the ones I will be discussing in class. Make your datagram payload a fixed 512 bytes, inclusive of the file transfer protocol header (which must, at the very least, carry: the sequence number of the datagram; ACKs; and advertised window notifications). The client reads the file contents in its receive buffer and prints them out on stdout using a separate thread. This thread sits in a repetitive loop till all the file contents have been printed out, doing the following. It samples from an exponential distribution with mean µ milliseconds (read from the client.in file), sleeps for that number of milliseconds; wakes up to read and print all in-order file contents available in the receive buffer at that point; samples again from the exponential distribution; sleeps; and so on. The formula -1 × µ × ln( random( ) ) , where ln is the natural logarithm, yields variates from an exponential distribution with mean µ, based on the uniformly-distributed variates over ( 0 , 1 ) returned by random(). Note that you will need to implement some sort of mutual exclusion/semaphore mechanism on the client side so that the thread that sleeps and wakes up to consume from the receive buffer is not updating the state variables of the buffer at the same time as the main thread reading from the socket and depositing into the buffer is doing the same. Furthermore, we need to ensure that the main thread does not effectively monopolize the semaphore (and thus lock out for prolonged periods of time) the sleeping thread when the latter wakes up. See the textbook, Section 26.7, ‘Mutexes: Mutual Exclusion’, pp.697-701. You might also find Section 26.8, ‘Condition Variables’, pp.701-705, useful. You will need to devise some way by which the sender can notify the receiver when it has sent the last datagram of the file transfer, without the receiver mistaking that EOF marker as part of the file contents. (Also, note that the last data segment could be a “short” segment of less than 512 bytes – your client needs to be able to handle this correctly somehow.) When the sender receives an ACK for the last datagram of the transfer, the (child) server terminates. The parent server has to take care of cleaning up zombie children. Note that if we want a clean closing, the client process cannot simply terminate when the receiver ACKs the last datagram. This ACK could be lost, which would leave the (child) server process ‘hanging’, timing out, and retransmitting the last datagram. TCP attempts to deal with this problem by means of the TIME_WAIT state. You should have your receiver process behave similarly, sticking around in something akin to a TIME_WAIT state in case in case it needs to retransmit the ACK. In the ReadMe file you hand in, report on how you dealt with the issues raised here: sender notifying receiver of the last datagram, clean closing, and so on. Output Some of the output required from your program has been described in the section Operation above. I expect you to provide further output – clear, well-structured, well-laid-out, concise but sufficient and helpful – in the client and server windows by means of which we can trace the correct evolution of your TCP’s behaviour in all its intricacies : information (e.g., sequence number) on datagrams and acks sent and dropped, window advertisements, datagram retransmissions (and why : dup acks or RTO); entering/exiting Slow Start and Congestion Avoidance, ssthresh and cwnd values; sender and receiver windows locking/unlocking; etc., etc. . . . . The onus is on you to convince us that the TCP mechanisms you implemented are working correctly. Too many students do not put sufficient thought, creative imagination, time or effort into this. It is not the TA’s nor my responsibility to sit staring at an essentially blank screen, trying to summon up our paranormal psychology skills to figure out if your TCP implementation is really working correctly in all its very intricate aspects, simply because the transferred file seems to be printing o.k. in the client window. Nor is it our responsibility to strain our eyes and our patience wading through a mountain of obscure, ill-structured, hyper-messy, debugging-style output because, for example, your effort-conserving concept of what is ‘suitable’ is to dump your debugging output on us, relevant, irrelevant, and everything in between.
nokia-t1zhou / Segment Routing Step By Stepsegment routing/segment routing v6 study and practice, include: SR-TE, SRv6 VPN
ebiken / P4srv6Proto-typing SRv6 functions with P4 lang.
averyvery / FreebieEE 2.0 extension for ignoring specific segments when routing
unixdigest / Openbsd Router GuideA guide for setting up an OpenBSD router with a network segmenting firewall, DHCP, DNS with Unbound, domain blocking and much more https://openbsdrouterguide.net/
Zashas / Thesis SRv6 BPFCode written for my master's thesis on End.BPF, an interface for programmable IPv6 Segment Routing network functions.
srl-labs / Nokia Segment Routing LabA lab to demonstrate Flex-Algo use case
SpaceNetLab / SpaceRTC ProjectUser-perceived latency is important for the quality of experience (QoE) of wide-area real-time communications (RTC). This paper explores a futuristic yet important problem facing the RTC community: can we exploit emerging mega-constellations to facilitate low-latency RTC globally? We carry out our quest in three steps. First, through a measurement study associated with a large number of geo-distributed RTC users, we quantitatively expose that the meandering routes over the client-cloud segment and the inter-cloud-site segment in existing cloud-based RTC architecture are critical culprits for the high latency issue suffered by wide-area RTC sessions. Second, after analyzing the low-latency potential enabled by mega-constellations, we formulate the dynamic RTC latency minimization problem under the integrated space-ground environment, and propose SpaceRTC, a satellite-cloud cooperative framework that adaptively selects relay servers upon satellites and cloud sites to build an overlay network which enables diverse close-to-optimal paths, and then judiciously allocates RTC flows in the network to facilitate low-latency interactions. Finally, we implement a hardware-in-the-loop experimental environment based on public constellation information and RTC trace, and extensive experiments demonstrate that SpaceRTC can deliver near-optimal interactive latency, with up to 64.9% latency reduction as compared with other state-of-the-art cloud-based solutions under representative videoconferencing traffic.
SOYJUN / Application With Raw IP SocketsOverview For this assignment you will be developing an application that uses raw IP sockets to ‘walk’ around an ordered list of nodes (given as a command line argument at the ‘source’ node, which is the node at which the tour was initiated), in a manner similar to the IP SSRR (Strict Source and Record Route) option. At each node, the application pings the preceding node in the tour. However, unlike the ping code in Stevens, you will be sending the ping ICMP echo request messages through a SOCK_RAW-type PF_PACKET socket and implementing ARP functionality to find the Ethernet address of the target node. Finally, when the ‘walk’ is completed, the group of nodes visited on the tour will exchange multicast messages. Your code will consist of two process modules, a ‘Tour’ application module (which will implement all the functionality outlined above, except for ARP activity) and an ARP module. The following should prove to be useful reference material for the assignment: Sections 21.2, 21.3, 21.6 and 21.10, Chapter 21, on Multicasting. Sections 27.1 to 27.3, Chapter 27, on the IP SSRR option. Sections 28.1 to 28.5, Chapter 28, on raw sockets, the IP_HDRINCL socket option, and ping. Sections 15.5, Chapter 15, on Unix domain SOCK_STREAM sockets. Figure 29.14, p. 807, and the corresponding explanation on p. 806, on filling in an IP header when the IP_HDRINCL socket option is in effect. The Lecture Slides on ARP & RARP (especially Section 4.4, ARP Packet Format, and the Figure 4.3 it includes). The link http://www.pdbuchan.com/rawsock/rawsock.html contains useful code samples that use IP raw sockets and PF_PACKET sockets. Note, in partcular, the code “icmp4_ll.c” in Table 2 for building an echo request sent through a PF_PACKET SOCK_RAW socket. The VMware environment You will be using the same vm1 , . . . . . , vm10 nodes you used for Assignment 3. However, unlike Assignment 3, you should use only interfaces eth0 and their associated IP addresses and ignore the other Ethernet interfaces that nodes have (interfaces eth0 make vm1 , . . . . . , vm10 look as if they belong to the same Ethernet LAN segment IP network 130.245.156.0/24). Note that, apart from the primary IP addresses associated with interfaces eth0, some nodes might also have one or more alias IP addresses associated with their interface eth0. Tour application module specifications The application will create a total of four sockets: two IP raw sockets, a PF_PACKET socket and a UDP socket for multicasting. We shall call the two IP raw sockets the ‘rt ’ (‘route traversal’) and ‘pg ’ (‘ping’) sockets, respectively. The rt socket should have the IP_HDRINCL option set. You will only be receiving ICMP echo reply messages through the pg socket (and not sending echo requests), so it does not matter whether it has the IP_HDRINCL option set or not. The pg socket should have protocol value (i.e., protocol demultiplexing key in the IP header) IPPROTO_ICMP. The rt socket should have a protocol value that identifies the application - i.e., some value other than the IPPROTO_XXXX values in /usr/include/netinet/in.h. However, remember that you will all be running your code using the same root account on the vm1 , . . . . . , vm10 nodes. So if two of you happen to choose the same protocol value and happen to be running on the same vm node at the same time, your applications will receive each other’s IP packets. For that reason, try to choose a protocol value for your rt socket that is likely to be unique to yourself. The PF_PACKET socket should be of type SOCK_RAW (not SOCK_DGRAM). This socket should have a protocol value of ETH_P_IP = 0x0800 (IPv4). The UDP socket for multicasting will be discussed below. Note that, depending on how you choose to bind that socket, you might actually need to have two UDP sockets for multicast communication – see bottom of p. 576, Section 21.10. Your application will, of course, have to be running on every vm node that is included in the tour. When evoking the application on the source node, the user supplies a sequence of vm node names (not IP addresses) to be visited in order. This command line sequence starts with the next node to be visited from the source node (i.e., it does not start with the source node itself). The sequence can include any number of repeated visits to the same node. For example, suppose that the source node is vm3 and the executable is called badr_tour : [root@vm3/root]# badr_tour vm2 vm10 vm4 vm7 vm5 vm2 vm6 vm2 vm9 vm4 vm7 vm2 vm6 vm5 vm1 vm10 vm8 (but note that the tour does not necessarily have to visit every vm node; and the same node should not appear consequentively in the tour list – i.e., the next node on the tour cannot be the current node itself). The application turns the sequence into a list of IP addresses for source routing. It also adds the IP address of the source node itself to the beginning of the list. The list thus produced will be carried as the payload of an IP packet, not as a SSRR option in the packet header. It is our application which will ensure that every node in the sequence is visited in order, not the IP SSRR capability. The source node should also add to the list an IP multicast address and a port number of its choice. It should also join the multicast group at that address and port number on its UDP socket. The TTL for outgoing multicasts should be set to 1. The application then fills in the header of an IP packet, designating itself as the IP source, and the next node to be visited as the IP destination. The packet is sent out on the rt socket. Note that on Linux, all the fields of the packet header must be in network byte order (Stevens, Section 28.3, p. 737, the fourth bullet point). When filling in the packet header, you should explicitly fill in the identification field (recall that, with the IP_HDRINCL socket option, if the identification field is given value 0, then the kernel will set its value). Try to make sure that the value you choose is likely to be unique to yourself (for reasons similar to those explained with respect to the IPPROTO_XXXX in 1. above). When a node receives an IP packet on its rt socket, it should first check that the identification field carries the right value (this implies that you will hard code your choice of identification field value determined in item 2 above in your code). If the identification field value does not check out, the packet is ignored. For a valid packet : Print out a message along the lines of: <time> received source routing packet from <hostname> <time> is the current time in human-readable format (see lines 19 & 20 in Figure 1.9, p. 14, and the corresponding explanation on p. 14f.), and <hostname> is the host name corresponding to the source IP address in the header of the received packet. If this is the first time the node is visited, the application should use the multicast address and port number in the packet received to join the multicast group on its UDP socket. The TTL for outgoing multicasts should be set to 1. The application updates the list in the payload, so that the next node in the tour can easily identify what the next hop from itself will be when it receives the packet. How you do this I leave up to you. You could, for example, include as part of the payload a pointer field into the list of nodes to be visited. This pointer would then be updated to the next entry in the list as the packet progresses hop by hop (see Figure 27.1 and the associated explanation on pp. 711-712). Other solutions are, of course, possible. The application then fills in a new IP header, designating itself as the IP source, and the next node to be visited as the IP destination. The identification field should be set to the same value as in the received packet. The packet is sent out on the rt socket. The node should also initiate pinging to the preceding node in the tour (the IP address of which it should pick up from the header of the received packet). However, unlike the Stevens ping code, it will be using the SOCK_RAW-type PF_PACKET socket of item 1 above to send the ICMP echo request messages. Before it can send echo request messages, the application has to call on the ARP module you will implement to get the Ethernet address of this preceding / ‘target’ node; this call is made using the API function areq which you will also implement (see sections ARP module specifications & API specifications below). Note that ARP has to be evoked every time the application wants to send out an echo request message, and not just the first time. An echo request message has to be encapsulated in a properly-formulated IP packet, which is in turn encapsulated in a properly-formulated Ethernet frame transmitted out through the PF_PACKET socket ; otherwise, ICMP at the source node will not receive it. You will have to modify Stevens’ ping code accordingly, specifically, the send_v4 function. In particular, the Ethernet frame must have a value of ETH_P_IP = 0x0800 (IPv4 – see <linux/if_ether.h>) in the frame type / ‘length’ field ; and the encapsulated IP packet must have a value of IPPROTO_ICMP = 0x01 (ICMPv4 – see <netinet_in.h>) in its protocol field. You should also simplify the ping code in its entirety by stripping all the ‘indirection’ IPv4 / IPv6 dual-operability paraphernalia and making the code work just for IPv4. Also note that the functions host_serv and freeaddrinfo, together with the associated structure addrinfo (see Sections 11.6, 11.8 & 11.11), in Figures 27.3, 27.6 & 28.5 ( pp. 713, 716 & 744f., respectively) can be replaced by the function gethostbyname and associated structure hostent (see Section 11.3) where needed. Also, there is no ‘-v’ verbose option, so this too should be stripped from Stevens’ code. When a node is ready to start pinging, it first prints out a ‘PING’ message similar to lines 32-33 of Figure 28.5, p. 744. It then builds up ICMP echo request messages and sends them to the source node every 1 second through the PF_PACKET socket. It also reads incoming echo response messages off the pg socket, in response to which it prints out the same kind of output as the code of Figure 28.8, p. 748. If this node and its preceding node have been previously visited in that order during the tour, then pinging would have already been initiated from the one to the other in response to the first visit, and nothing further should nor need be done during second and subsequent visits. In light of the above, note that once a node initiates pinging, it needs to read from both its rt and pg sockets, necessitating the use of the select function. As will be clear from what follows below, the application will anyway be needing also to simultaneously monitor its UDP socket for incoming multicast datagrams. When the last node on the tour is reached, and if this is the first time it is visited, it joins the multicast group and starts pinging the preceding node (if it is not already doing so). After a few echo replies are received (five, say), it sends out the multicast message below on its UDP socket (i.e., the node should wait about five seconds before sending the multicast message) : <<<<< This is node vmi . Tour has ended . Group members please identify yourselves. >>>>> where vmi is the name (not IP address) of the node. The node should also print this message out on stdout preceded, on the same line, by the phrase: Node vmi . Sending: <then print out the message sent>. Each node vmj receiving this message should print out the message received preceded, on the same line, by the phrase: Node vmj . Received <then print out the message received>. Each such node in step a above should then immediately stop its pinging activity. The node should then send out the following multicast message: <<<<< Node vmj . I am a member of the group. >>>>> and print out this message preceded, on the same line, by the phrase: Node vmj . Sending: <then print out the message sent>. Each node receiving these second multicast messages (i.e., the messages that nodes – including itself – sent out in step c above) should print each such message out preceded, on the same line, by the phrase: Node vmk . Received: <then print out the message received>. Reading from the socket in step d above should be implemented with a 5-second timeout. When the timeout expires, the node should print out another message to the effect that it is terminating the Tour application, and gracefully exit its Tour process. Note that under Multicast specifications, the last node in the tour, which sends out the End of Tour message, should itself receive a copy of that message and, when it does, it should behave exactly as do the other nodes in steps a. – e. above. ARP module specifications Your executable is evoked with no command line arguments. Like the Tour module, it will be running on every vm node. It uses the get_hw_addrs function of Assignment 3 to explore its node’s interfaces and build a set of <IP address , HW address> matching pairs for all eth0 interface IP addresses (including alias IP addresses, if any). Write out to stdout in some appropriately clear format the address pairs found. The module creates two sockets: a PF_PACKET socket and a Unix domain socket. The PF_PACKET should be of type SOCK_RAW (not type SOCK_DGRAM) with a protocol value of your choice (but not one of the standard values defined in <linux/if_ether.h>) which is, hopefully, unique to yourself. This value effectively becomes the protocol value for your implementation of ARP. Because this protocol value will be carried in the frame type / ‘length’ field of the Ethernet frame header (see Figure 4.3 of the ARP & RARP handout), the value chosen should be not less than 1536 (0x600) so that it is not misinterpreted as the length of an Ethernet 802.3 frame. The Unix domain socket should be of type SOCK_STREAM (not SOCK_DGRAM). It is a listening socket bound to a ‘well-known’ sun_path file. This socket will be used to communicate with the function areq that is implemented in the Tour module (see the section API specifications below). In this context, areq will act as the client and the ARP module as the server. The ARP module then sits in an infinite loop, monitoring these two sockets. As ARP request messages arrive on the PF_PACKET socket, the module processes them, and responds with ARP reply messages as appropriate. The protocol builds a ‘cache’ of matching <IP address , HW address> pairs from the replies (and requests – see below) it receives. For simplicity, and unlike the real ARP, we shall not implement timing out mechanisms for these cache entries. A cache entry has five parts: (i) IP address ; (ii) HW address ; (iii) sll_ifindex (the interface to be used for reaching the matching pair <(i) , (ii)>) ; (iv) sll_hatype ; and (v) a Unix-domain connection-socket descriptor for a connected client (see the section API specifications below for the latter three). When an ARP reply is being entered in the cache, the ARP module uses the socket descriptor in (v) to send a reply to the client, closes the connection socket, and deletes the socket descriptor from the cache entry. Note that, like the real ARP, when an ARP request is received by a node, and if the request pertains to that receiving node, the sender’s (see Figure 4.3 of the ARP & RARP handout) <IP address, HW address> matching pair should be entered into the cache if it is not already there (together, of course, with (iii) sll_ifindex & (iv) sll_hatype), or updated if need be if such an entry already exists in the cache. If the ARP request received does not pertain to the node receiving it, but there is already an entry in that receiving node's cache for the sender’s <IP address, HW address> matching pair, that entry should be checked and updated if need be. If there is no such entry, no action is taken (in particular, and unlike the case above, no new entry should be made in the receiving node's cache of the sender’s <IP address, HW address> matching pair if such an entry does not already exist). ARP request and reply messages have the same format as Figure 4.3 of the ARP & RARP handout, but with an extra 2-byte identification field added at the beginning which you fill with a value chosen so that it has a high probability of being unique to yourself. This value is to be echoed in the reply message, and helps to act as a further filter in case some other student happens to have fortuitously chosen the same value as yourself for the protocol parameter of the ARP PF_PACKET. Values in the fields of our ARP messages must be in network byte order. You might find the system header file <linux/if_arp.h> useful for manipulating ARP request and reply messages, but remember that our version of these messages have an extra two-byte field as mentioned above. Your code should print out on stdout, in some appropriately clear format, the contents of the Ethernet frame header and ARP request message you send. As described in Section 4.4 of the ARP & RARP handout, the node that responds to the request should, in its reply message, swap the two sender addresses with the two target addresses, as well as, of course, echo back the extra identification field sent with the request. The protocol at this responding node should print out, in an appropriately clear format, both the request frame (header and ARP message) it receives and the reply frame it sends. Similarly, the node that sent the request should print out the reply frame it receives. Finally, recall that the node issuing the request sends out a broadcast Ethernet frame, but the responding node replies with a unicast frame. API specifications The API is for communication between the Tour process and the ARP process. It consists of a single function, areq, implemented in the Tour module. areq is called by send_v4 function of the application every time the latter want to send out an ICMP echo request message: int areq (struct sockaddr *IPaddr, socklen_t sockaddrlen, struct hwaddr *HWaddr); IPaddr contains the primary or alias IPaddress of a ‘target’ node on the LAN for which the corresponding hardware address is being requested. hwaddr is a new structure (and not a pre-existing type) modeled on the sockaddr_ll of PF_PACKET; you will have to declare it in your code. It is used to return the requested hardware address to the caller of areq : structure hwaddr { int sll_ifindex; /* Interface number */ unsigned short sll_hatype; /* Hardware type */ unsigned char sll_halen; /* Length of address */ unsigned char sll_addr[8]; /* Physical layer address */ }; areq creates a Unix domain socket of type SOCK_STREAM and connects to the ‘well-known’ sun_path file of the ARP listening socket. It sends the IP address from parameter IPaddr and the information in the three fields of parameter HWaddr to ARP. It then blocks on a read awaiting a reply from ARP. This read should be backed up by a timeout since it is possible that no reply is received for the request. If a timeout occurs, areq should close the socket and return to its caller indicating failure (through its int return value). Your application code should print out on stdout, in some appropriately clear format, a notification every time areq is called, giving the IP address for which a HW address is being sought. It should similarly print out the result when the call to areq returns (HW address returned, or failure). When the ARP module receives a request for a HW address from areq through its Unix domain listening socket, it first checks if the required HW address is already in the cache. If so, it can respond immediately to the areq and close the Unix domain connection socket. Else : it makes an ‘incomplete’ entry in the cache, consisting of parts (i), (iii), (iv) and (v) ; puts out an ARP request message on the network on its PF_PACKET socket; and starts monitoring the areq connection socket for readability – if the areq client closes the connection socket (this would occur in response to a timeout in areq), ARP deletes the corresponding incomplete entry from the cache (and ignores any subsequent ARP reply from the network if such is received). On the other hand, if ARP receives a reply from the network, it updates the incomplete cache entry, responds to areq, and closes the connection socket.
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