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• Linux bonded Interfaces hanging periodically

  - by David

  I've several hosts that are showing problems with connectivity. When working from the command line, for example, typing is frozen for a second or so, then recovers - then it does it again. The most egregious example host would freeze (input) for 15-30 seconds, then recover and go out 5 seconds later. Switching cables didn't do anything - but removing one of the physical cables caused everything to clear up instantly (which why I think this is a network problem). Looking at the network I couldn't see any packets floating that would explain this. These ethernet interfaces (Gigabit Dell) were working normally previously, but since we moved the systems - and put them on a new set of switches - this has been a problem on multiple theoretically identically-configured hosts. The original switches were an HP Procurve 1810-24G and an HP Procurve 1800-24G connected with LLDP; the new switches are both Cisco SG 200-26, which I understand are rebranded Linksys switches. Is this caused by a problem with the switches? Is it the switch configurations? Are the Cisco switches incapable of handling this? I don't see where the configuration is located; I searched the usual /etc/sysconfig/network/devices but there's nothing in there about options (like mii polling) and nothing about the method of balancing the two. Searching scripts, I can't find anything in /etc/init.d/network either. The hosts are almost all Red Hat Enterprise Linux 5.x systems (5.6, 5.7) but some are Ubuntu Server 10.04.3 Lucid Lynx. I need help with both if it comes to that. UPDATE: We're also seeing some problems with servers on the original switches. The HP switches and the Cisco switches are also interconnected (temporarily); there is a cable run from one switch to the next. Pings on any of these hosts show about one ICMP packet out of every 5-6 getting dropped (timed out). Could there be an interaction between the two switches? Oh, and the hosts are using bonding with Balance-RR as the method.

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• Is this iptables NAT exploitable from the external side?

  - by Karma Fusebox

  Could you please have a short look on this simple iptables/NAT-Setup, I believe it has a fairly serious security issue (due to being too simple). On this network there is one internet-connected machine (running Debian Squeeze/2.6.32-5 with iptables 1.4.8) acting as NAT/Gateway for the handful of clients in 192.168/24. The machine has two NICs: eth0: internet-faced eth1: LAN-faced, 192.168.0.1, the default GW for 192.168/24 Routing table is two-NICs-default without manual changes: Destination Gateway Genmask Flags Metric Ref Use Iface 192.168.0.0 0.0.0.0 255.255.255.0 U 0 0 0 eth1 (externalNet) 0.0.0.0 255.255.252.0 U 0 0 0 eth0 0.0.0.0 (externalGW) 0.0.0.0 UG 0 0 0 eth0 The NAT is then enabled only and merely by these actions, there are no more iptables rules: echo 1 > /proc/sys/net/ipv4/ip_forward /sbin/iptables -t nat -A POSTROUTING -o eth0 -j MASQUERADE # (all iptables policies are ACCEPT) This does the job, but I miss several things here which I believe could be a security issue: there is no restriction about allowed source interfaces or source networks at all there is no firewalling part such as: (set policies to DROP) /sbin/iptables -A FORWARD -i eth0 -o eth1 -m state --state RELATED,ESTABLISHED -j ACCEPT /sbin/iptables -A FORWARD -i eth1 -o eth0 -j ACCEPT And thus, the questions of my sleepless nights are: Is this NAT-service available to anyone in the world who sets this machine as his default gateway? I'd say yes it is, because there is nothing indicating that an incoming external connection (via eth0) should be handled any different than an incoming internal connection (via eth1) as long as the output-interface is eth0 - and routing-wise that holds true for both external und internal clients that want to access the internet. So if I am right, anyone could use this machine as open proxy by having his packets NATted here. So please tell me if that's right or why it is not. As a "hotfix" I have added a "-s 192.168.0.0/24" option to the NAT-starting command. I would like to know if not using this option was indeed a security issue or just irrelevant thanks to some mechanism I am not aware of. As the policies are all ACCEPT, there is currently no restriction on forwarding eth1 to eth0 (internal to external). But what are the effective implications of currently NOT having the restriction that only RELATED and ESTABLISHED states are forwarded from eth0 to eth1 (external to internal)? In other words, should I rather change the policies to DROP and apply the two "firewalling" rules I mentioned above or is the lack of them not affecting security? Thanks for clarification!

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• VPC SSH port forward into private subnet

  - by CP510

  Ok, so I've been racking my brain for DAYS on this dilema. I have a VPC setup with a public subnet, and a private subnet. The NAT is in place of course. I can connect from SSH into a instance in the public subnet, as well as the NAT. I can even ssh connect to the private instance from the public instance. I changed the SSHD configuration on the private instance to accept both port 22 and an arbitrary port number 1300. That works fine. But I need to set it up so that I can connect to the private instance directly using the 1300 port number, ie. ssh -i keyfile.pem [email protected] -p 1300 and 1.2.3.4 should route it to the internal server 10.10.10.10. Now I heard iptables is the job for this, so I went ahead and researched and played around with some routing with that. These are the rules I have setup on the public instance (not the NAT). I didn't want to use the NAT for this since AWS apperantly pre-configures the NAT instances when you set them up and I heard using iptables can mess that up. *filter :INPUT ACCEPT [129:12186] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [84:10472] -A INPUT -i lo -j ACCEPT -A INPUT -i eth0 -p tcp -m state --state NEW -m tcp --dport 1300 -j ACCEPT -A INPUT -d 10.10.10.10/32 -p tcp -m limit --limit 5/min -j LOG --log-prefix "SSH Dropped: " -A FORWARD -d 10.10.10.10/32 -p tcp -m tcp --dport 1300 -j ACCEPT -A OUTPUT -o lo -j ACCEPT COMMIT # Completed on Wed Apr 17 04:19:29 2013 # Generated by iptables-save v1.4.12 on Wed Apr 17 04:19:29 2013 *nat :PREROUTING ACCEPT [2:104] :INPUT ACCEPT [2:104] :OUTPUT ACCEPT [6:681] :POSTROUTING ACCEPT [7:745] -A PREROUTING -i eth0 -p tcp -m tcp --dport 1300 -j DNAT --to-destination 10.10.10.10:1300 -A POSTROUTING -p tcp -m tcp --dport 1300 -j MASQUERADE COMMIT So when I try this from home. It just times out. No connection refused messages or anything. And I can't seem to find any log messages about dropped packets. My security groups and ACL settings allow communications on these ports in both directions in both subnets and on the NAT. I'm at a loss. What am I doing wrong?

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• Wrapping a point-to-point link

  - by user3712955

  I'm using a pair of IP radios (non-WiFi) to bridge my office engineering LAN (172.0.0.0/8) to a lab in another building. The radios work fine, but they expose a web management interface I'd like to hide, and they also generate traffic (ARP, STP, and more) that I need to keep off my (very, very clean) LAN segments. I have some ARM-Linux boards (similar to Beagle/Panda/RasPi) running Ubuntu, and I've put one at each end of the link, between the radio and the LAN. Each of the boards has 2 wired Ethernet interfaces, eth0 and eth1. The LAN segments are connected to eth0, and the radios are connected to eth1. I'd like to accomplish the following: Keep radio-originated traffic off my LAN segments! Hide all services provided by the radio (web, ssh, etc.) Transparently pass all traffic between the LAN segments (including things like ARP). The above also applies to the ARM-Linux boards: No stray traffic my LAN from them either! I'd like the system to look like a switch: LAN packets arriving at one eth0 appear at the other. And neither eth0 should have an IP address: The working system should behave like a CAT6 cable with some latency (instead of ARM boards and radios). Unfortunately, I'm confused about how to properly configure the ARM Ubuntu systems. What I'm guessing I need is a bridge on each board (br0?) and a VLAN (vlan0 or eth0.0?) to isolate the LAN traffic from everything else as it passes through the ARM boards and the radios. Then I need some kind of a firewall to block sending anything out eth0 that isn't from the other eth0 (via the VLAN). I've looked at the ip and ebtables commands (especially -t broute). While the concepts sorta-kinda make sense, I'm completely lost in the details. Edit: In the perverse case that a system on one of my LAN segments has the same IP address as one of the radios, or as eth1 on the ARM-Ubuntu boards, a VLAN won't work. Which I believe means I need to tunnel all traffic between the two eth0 interfaces to get that "like a wire" behavior. Help? Finally, I'd like to have a way to temporarily expose services on the ARM boards (ssh) and the radios (web) for maintenance purposes. Ideally, it would expose an IP address with ssh available on port 22. Once connected, I figure I'd start an X11 session and run a browser on the ARM board to access the radios. Or something. I could login via the console to enable/disable this, or perhaps could use a GPIO to trigger a script. I feel I've identified most of the pieces needed to make all this happen, but I have no idea how to combine them to make a working system. Thanks!

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• Internet Pings but Does Not Load

  - by t3techcom18

  From what I've been seeing and been doing my research for the past two days, many people have been having the same issues throughout the years, however, this is the first time I've encountered this issue and many of the specific workarounds or fixes have not worked for me. I've been trying to work through this for 24 hours straight now, but to no avail so many thanks to those that can help. On Monday night, got home from work; surfing the internet for half an hour, everything was fine as always. Just after half an hour, my Internet got very sluggish and then it died completely. I thought it might have been the an update I just put through in terms of Windows Update that said was a critical update for MSE, as the same thing happened a few years ago. I did a System Restore to two different dates that were in the past two weeks, nothing. Uninstalled MSE and disabled Windows Defender and the Windows Firewall: Nothing. Reset IE Options, Reset Winsock, Dumping DNS, many of the other command prompt screens to reset items: Nothing. Reset the modem: Nothing. What DID work, however, was a ping test to Yahoo. The ping test worked, saying all four packets was recieved, yet nothing else popped up. LAN and CenturyLink said everything worked on their end and that everything was connected properly, as well as the speeds working fine. CenturyLink said in their notes that they thought Port 80 was blocked. I went and put in the Firewall to allow Port 80 but it didn't make any difference whatsoever. I remembered I had a spare modem laying around and I switched them up, both modem and the cords - nothing. I then hooked it up to my netbook to see if that would work, as it usually does - connection didn't work there either. Like I said, it's been about 24 hours now and this is increasingly frustrating, as I've tried all solutions (While browsing through 10 search results pages on my phone) suggested and still nothing. Any suggestions and tricks would be greatly appreciated! Here's my specs: Windows 7 32-bit Home Premium Intel Core 2 Duo 3.14 Ghz 4 GB Kingston DDR2 RAM eVGA nForce 750i SLI eVGA GeForce GTX 560 Ti FPB ISP: CenturyLink No router Modem: CenturyLink 660 Series Hardwired connection PLEASE NOTE: This is the only computer I have (Like I said, the netbook solution didn't work), so downloading programs and such is not an option til I get to other computers somewhere else, like right now. Unless someone knows of a way of copying/pasting a file in Windows and then transferring said info to an Android smartphone, this is gunna take a while haha. Patience is requested.

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• ERROR: Linux route add command failed: external program exited with error status: 4

  - by JohnMerlino

  A remote machine running fedora uses openvpn, and multiple developers were successfully able to connect to it via their client openvpn. However, I am running Ubuntu 12.04 and I am having trouble connecting to the server via vpn. I copied ca.crt, home.key, and home.crt from the server to my local machine to /etc/openvpn folder. My client.conf file looks like this: ############################################## # Sample client-side OpenVPN 2.0 config file # # for connecting to multi-client server. # # # # This configuration can be used by multiple # # clients, however each client should have # # its own cert and key files. # # # # On Windows, you might want to rename this # # file so it has a .ovpn extension # ############################################## # Specify that we are a client and that we # will be pulling certain config file directives # from the server. client # Use the same setting as you are using on # the server. # On most systems, the VPN will not function # unless you partially or fully disable # the firewall for the TUN/TAP interface. ;dev tap dev tun # Windows needs the TAP-Win32 adapter name # from the Network Connections panel # if you have more than one. On XP SP2, # you may need to disable the firewall # for the TAP adapter. ;dev-node MyTap # Are we connecting to a TCP or # UDP server? Use the same setting as # on the server. ;proto tcp proto udp # The hostname/IP and port of the server. # You can have multiple remote entries # to load balance between the servers. remote xx.xxx.xx.130 1194 ;remote my-server-2 1194 # Choose a random host from the remote # list for load-balancing. Otherwise # try hosts in the order specified. ;remote-random # Keep trying indefinitely to resolve the # host name of the OpenVPN server. Very useful # on machines which are not permanently connected # to the internet such as laptops. resolv-retry infinite # Most clients don't need to bind to # a specific local port number. nobind # Downgrade privileges after initialization (non-Windows only) ;user nobody ;group nogroup # Try to preserve some state across restarts. persist-key persist-tun # If you are connecting through an # HTTP proxy to reach the actual OpenVPN # server, put the proxy server/IP and # port number here. See the man page # if your proxy server requires # authentication. ;http-proxy-retry # retry on connection failures ;http-proxy [proxy server] [proxy port #] # Wireless networks often produce a lot # of duplicate packets. Set this flag # to silence duplicate packet warnings. ;mute-replay-warnings # SSL/TLS parms. # See the server config file for more # description. It's best to use # a separate .crt/.key file pair # for each client. A single ca # file can be used for all clients. ca ca.crt cert home.crt key home.key # Verify server certificate by checking # that the certicate has the nsCertType # field set to "server". This is an # important precaution to protect against # a potential attack discussed here: # http://openvpn.net/howto.html#mitm # # To use this feature, you will need to generate # your server certificates with the nsCertType # field set to "server". The build-key-server # script in the easy-rsa folder will do this. ns-cert-type server # If a tls-auth key is used on the server # then every client must also have the key. ;tls-auth ta.key 1 # Select a cryptographic cipher. # If the cipher option is used on the server # then you must also specify it here. ;cipher x # Enable compression on the VPN link. # Don't enable this unless it is also # enabled in the server config file. comp-lzo # Set log file verbosity. verb 3 # Silence repeating messages ;mute 20 But when I start server and look in /var/log/syslog, I notice the following error: May 27 22:13:51 myuser ovpn-client[5626]: /sbin/route add -net 10.27.12.1 netmask 255.255.255.252 gw 10.27.12.37 May 27 22:13:51 myuser ovpn-client[5626]: ERROR: Linux route add command failed: external program exited with error status: 4 May 27 22:13:51 myuser ovpn-client[5626]: /sbin/route add -net 172.27.12.0 netmask 255.255.255.0 gw 10.27.12.37 May 27 22:13:51 myuser ovpn-client[5626]: /sbin/route add -net 10.27.12.1 netmask 255.255.255.255 gw 10.27.12.37 And I am unable to connect to the server via openvpn: $ ssh [email protected] ssh: connect to host xxx.xx.xx.130 port 22: No route to host What may I be doing wrong?

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• Sun Fire X4800 M2 Delivers World Record TPC-C for x86 Systems

  - by Brian

  Oracle's Sun Fire X4800 M2 server equipped with eight 2.4 GHz Intel Xeon Processor E7-8870 chips obtained a result of 5,055,888 tpmC on the TPC-C benchmark. This result is a world record for x86 servers. Oracle demonstrated this world record database performance running Oracle Database 11g Release 2 Enterprise Edition with Partitioning. The Sun Fire X4800 M2 server delivered a new x86 TPC-C world record of 5,055,888 tpmC with a price performance of $0.89/tpmC using Oracle Database 11g Release 2. This configuration is available 06/26/12. The Sun Fire X4800 M2 server delivers 3.0x times better performance than the next 8-processor result, an IBM System p 570 equipped with POWER6 processors. The Sun Fire X4800 M2 server has 3.1x times better price/performance than the 8-processor 4.7GHz POWER6 IBM System p 570. The Sun Fire X4800 M2 server has 1.6x times better performance than the 4-processor IBM x3850 X5 system equipped with Intel Xeon processors. This is the first TPC-C result on any system using eight Intel Xeon Processor E7-8800 Series chips. The Sun Fire X4800 M2 server is the first x86 system to get over 5 million tpmC. The Oracle solution utilized Oracle Linux operating system and Oracle Database 11g Enterprise Edition Release 2 with Partitioning to produce the x86 world record TPC-C benchmark performance. Performance Landscape Select TPC-C results (sorted by tpmC, bigger is better) System p/c/t tpmC Price/tpmC Avail Database MemorySize Sun Fire X4800 M2 8/80/160 5,055,888 0.89 USD 6/26/2012 Oracle 11g R2 4 TB IBM x3850 X5 4/40/80 3,014,684 0.59 USD 7/11/2011 DB2 ESE 9.7 3 TB IBM x3850 X5 4/32/64 2,308,099 0.60 USD 5/20/2011 DB2 ESE 9.7 1.5 TB IBM System p 570 8/16/32 1,616,162 3.54 USD 11/21/2007 DB2 9.0 2 TB p/c/t - processors, cores, threads Avail - availability date Oracle and IBM TPC-C Response times System tpmC Response Time (sec) New Order 90th% Response Time (sec) New Order Average Sun Fire X4800 M2 5,055,888 0.210 0.166 IBM x3850 X5 3,014,684 0.500 0.272 Ratios - Oracle Better 1.6x 1.4x 1.3x Oracle uses average new order response time for comparison between Oracle and IBM. Graphs of Oracle's and IBM's response times for New-Order can be found in the full disclosure reports on TPC's website TPC-C Official Result Page. Configuration Summary and Results Hardware Configuration: Server Sun Fire X4800 M2 server 8 x 2.4 GHz Intel Xeon Processor E7-8870 4 TB memory 8 x 300 GB 10K RPM SAS internal disks 8 x Dual port 8 Gbs FC HBA Data Storage 10 x Sun Fire X4270 M2 servers configured as COMSTAR heads, each with 1 x 3.06 GHz Intel Xeon X5675 processor 8 GB memory 10 x 2 TB 7.2K RPM 3.5" SAS disks 2 x Sun Storage F5100 Flash Array storage (1.92 TB each) 1 x Brocade 5300 switches Redo Storage 2 x Sun Fire X4270 M2 servers configured as COMSTAR heads, each with 1 x 3.06 GHz Intel Xeon X5675 processor 8 GB memory 11 x 2 TB 7.2K RPM 3.5" SAS disks Clients 8 x Sun Fire X4170 M2 servers, each with 2 x 3.06 GHz Intel Xeon X5675 processors 48 GB memory 2 x 300 GB 10K RPM SAS disks Software Configuration: Oracle Linux (Sun Fire 4800 M2) Oracle Solaris 11 Express (COMSTAR for Sun Fire X4270 M2) Oracle Solaris 10 9/10 (Sun Fire X4170 M2) Oracle Database 11g Release 2 Enterprise Edition with Partitioning Oracle iPlanet Web Server 7.0 U5 Tuxedo CFS-R Tier 1 Results: System: Sun Fire X4800 M2 tpmC: 5,055,888 Price/tpmC: 0.89 USD Available: 6/26/2012 Database: Oracle Database 11g Cluster: no New Order Average Response: 0.166 seconds Benchmark Description TPC-C is an OLTP system benchmark. It simulates a complete environment where a population of terminal operators executes transactions against a database. The benchmark is centered around the principal activities (transactions) of an order-entry environment. These transactions include entering and delivering orders, recording payments, checking the status of orders, and monitoring the level of stock at the warehouses. Key Points and Best Practices Oracle Database 11g Release 2 Enterprise Edition with Partitioning scales easily to this high level of performance. COMSTAR (Common Multiprotocol SCSI Target) is the software framework that enables an Oracle Solaris host to serve as a SCSI Target platform. COMSTAR uses a modular approach to break the huge task of handling all the different pieces in a SCSI target subsystem into independent functional modules which are glued together by the SCSI Target Mode Framework (STMF). The modules implementing functionality at SCSI level (disk, tape, medium changer etc.) are not required to know about the underlying transport. And the modules implementing the transport protocol (FC, iSCSI, etc.) are not aware of the SCSI-level functionality of the packets they are transporting. The framework hides the details of allocation providing execution context and cleanup of SCSI commands and associated resources and simplifies the task of writing the SCSI or transport modules. Oracle iPlanet Web Server middleware is used for the client tier of the benchmark. Each web server instance supports more than a quarter-million users while satisfying the response time requirement from the TPC-C benchmark. See Also Oracle Press Release -- Sun Fire X4800 M2 TPC-C Executive Summary tpc.org Complete Sun Fire X4800 M2 TPC-C Full Disclosure Report tpc.org Transaction Processing Performance Council (TPC) Home Page Ideas International Benchmark Page Sun Fire X4800 M2 Server oracle.com OTN Oracle Linux oracle.com OTN Oracle Solaris oracle.com OTN Oracle Database 11g Release 2 Enterprise Edition oracle.com OTN Sun Storage F5100 Flash Array oracle.com OTN Disclosure Statement TPC Benchmark C, tpmC, and TPC-C are trademarks of the Transaction Processing Performance Council (TPC). Sun Fire X4800 M2 (8/80/160) with Oracle Database 11g Release 2 Enterprise Edition with Partitioning, 5,055,888 tpmC, $0.89 USD/tpmC, available 6/26/2012. IBM x3850 X5 (4/40/80) with DB2 ESE 9.7, 3,014,684 tpmC, $0.59 USD/tpmC, available 7/11/2011. IBM x3850 X5 (4/32/64) with DB2 ESE 9.7, 2,308,099 tpmC, $0.60 USD/tpmC, available 5/20/2011. IBM System p 570 (8/16/32) with DB2 9.0, 1,616,162 tpmC, $3.54 USD/tpmC, available 11/21/2007. Source: http://www.tpc.org/tpcc, results as of 7/15/2011.

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• Computer Networks UNISA - Chap 15 &ndash; Network Management

  - by MarkPearl

  After reading this section you should be able to Understand network management and the importance of documentation, baseline measurements, policies, and regulations to assess and maintain a network’s health. Manage a network’s performance using SNMP-based network management software, system and event logs, and traffic-shaping techniques Identify the reasons for and elements of an asset managements system Plan and follow regular hardware and software maintenance routines Fundamentals of Network Management Network management refers to the assessment, monitoring, and maintenance of all aspects of a network including checking for hardware faults, ensuring high QoS, maintaining records of network assets, etc. Scope of network management differs depending on the size and requirements of the network. All sub topics of network management share the goals of enhancing the efficiency and performance while preventing costly downtime or loss. Documentation The way documentation is stored may vary, but to adequately manage a network one should at least record the following… Physical topology (types of LAN and WAN topologies – ring, star, hybrid) Access method (does it use Ethernet 802.3, token ring, etc.) Protocols Devices (Switches, routers, etc) Operating Systems Applications Configurations (What version of operating system and config files for serve / client software) Baseline Measurements A baseline is a report of the network’s current state of operation. Baseline measurements might include the utilization rate for your network backbone, number of users logged on per day, etc. Baseline measurements allow you to compare future performance increases or decreases caused by network changes or events with past network performance. Obtaining baseline measurements is the only way to know for certain whether a pattern of usage has changed, or whether a network upgrade has made a difference. There are various tools available for measuring baseline performance on a network. Policies, Procedures, and Regulations Following rules helps limit chaos, confusion, and possibly downtime. The following policies and procedures and regulations make for sound network management. Media installations and management (includes designing physical layout of cable, etc.) Network addressing policies (includes choosing and applying a an addressing scheme) Resource sharing and naming conventions (includes rules for logon ID’s) Security related policies Troubleshooting procedures Backup and disaster recovery procedures In addition to internal policies, a network manager must consider external regulatory rules. Fault and Performance Management After documenting every aspect of your network and following policies and best practices, you are ready to asses you networks status on an on going basis. This process includes both performance management and fault management. Network Management Software To accomplish both fault and performance management, organizations often use enterprise-wide network management software. There various software packages that do this, each collect data from multiple networked devices at regular intervals, in a process called polling. Each managed device runs a network management agent. So as not to affect the performance of a device while collecting information, agents do not demand significant processing resources. The definition of a managed devices and their data are collected in a MIB (Management Information Base). Agents communicate information about managed devices via any of several application layer protocols. On modern networks most agents use SNMP which is part of the TCP/IP suite and typically runs over UDP on port 161. Because of the flexibility and sophisticated network management applications are a challenge to configure and fine-tune. One needs to be careful to only collect relevant information and not cause performance issues (i.e. pinging a device every 5 seconds can be a problem with thousands of devices). MRTG (Multi Router Traffic Grapher) is a simple command line utility that uses SNMP to poll devices and collects data in a log file. MRTG can be used with Windows, UNIX and Linux. System and Event Logs Virtually every condition recognized by an operating system can be recorded. This is typically done using event logs. In Windows there is a GUI event log viewer. Similar information is recorded in UNIX and Linux in a system log. Much of the information collected in event logs and syslog files does not point to a problem, even if it is marked with a warning so it is important to filter your logs appropriately to reduce the noise. Traffic Shaping When a network must handle high volumes of network traffic, users benefit from performance management technique called traffic shaping. Traffic shaping involves manipulating certain characteristics of packets, data streams, or connections to manage the type and amount of traffic traversing a network or interface at any moment. Its goals are to assure timely delivery of the most important traffic while offering the best possible performance for all users. Several types of traffic prioritization exist including prioritizing traffic according to any of the following characteristics… Protocol IP address User group DiffServr VLAN tag in a Data Link layer frame Service or application Caching In addition to traffic shaping, a network or host might use caching to improve performance. Caching is the local storage of frequently needed files that would otherwise be obtained from an external source. By keeping files close to the requester, caching allows the user to access those files quickly. The most common type of caching is Web caching, in which Web pages are stored locally. To an ISP, caching is much more than just convenience. It prevents a significant volume of WAN traffic, thus improving performance and saving money. Asset Management Another key component in managing networks is identifying and tracking its hardware. This is called asset management. The first step to asset management is to take an inventory of each node on the network. You will also want to keep records of every piece of software purchased by your organization. Asset management simplifies maintaining and upgrading the network chiefly because you know what the system includes. In addition, asset management provides network administrators with information about the costs and benefits of certain types of hardware or software. Change Management Networks are always in a stage of flux with various aspects including… Software changes and patches Client Upgrades Shared Application Upgrades NOS Upgrades Hardware and Physical Plant Changes Cabling Upgrades Backbone Upgrades For a detailed explanation on each of these read the textbook (Page 750 – 761)

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• Android stream to Wowza

  - by Curtis Kiu

  I feel very confused about Android streaming to wowza. I am doing a video conference using rtmp cross-platform, but Android doesn't eat RTMP. Therefore I need to find another way to do it. Upstreaming I found a new open-source app called spydroid-ipcamera. It is using rtp, sending udp packets to computer, and opens it in vlc using the following sdp v=0 s=Unnamed m=video 5006 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1;profile-level-id=420016;sprop-parameter-sets=Z0IAFukBQHsg,aM4BDyA=; But it can't work. Then I follow wowza tutorial and stream to it and then play again in VLC. That works! I wrote it in http://code.google.com/p/spydroid-ipcamera/issues/detail?id=2 However when I want to add audio in the packet, it fails to work. I change to code in http://code.google.com/p/spydroid-ipcamera/source/browse/trunk/src/net/mkp/spydroid/CameraStreamer.java mr.setAudioSource(MediaRecorder.AudioSource.MIC); mr.setVideoSource(MediaRecorder.VideoSource.CAMERA); mr.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4); mr.setVideoFrameRate(20); mr.setVideoSize(640, 480); mr.setAudioEncoder(MediaRecorder.AudioEncoder.AAC); mr.setVideoEncoder(MediaRecorder.VideoEncoder.H264); mr.setPreviewDisplay(holder.getSurface()); Then I thought that the problem should be in sdp, but I don't know how to due with sdp. I am streaming H.264/AAC with Mp4 Second I don't understand sdp. So how can I make video conference upstreaming part using this apps. Android ----(UDP Port:5006)----> PC (SDP file) and then Wowza read the SDP file ------> VLC I think in this way the system cannot handle more than 1 client. sdp can only hold 1 port, any idea or actually it wont' work? Also Wowza need to set the stream before we stream it, so does it mean that I should not follow this way to do it? Sorry my English is poor, I hope you guys understand.

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• C# listview add items

  - by Eyla

  Greetings, I have a method to capture packets. after capturing I want to add each packet as a row in listview in runting time so as soon as I capture packet I want to add it to the list view. the problem that when I use items.Add() I will get overload argument error. please advice!! here is my code: private void packetCapturingThreadMethod() { Packet packet = null; int countOfPacketCaptures = 0; while ((packet = device.GetNextPacket()) != null) { packet = device.GetNextPacket(); if (packet is TCPPacket) { TCPPacket tcp = (TCPPacket)packet; myPacket tempPacket = new myPacket(); tempPacket.packetType = "TCP"; tempPacket.sourceAddress = Convert.ToString(tcp.SourceAddress); tempPacket.destinationAddress = Convert.ToString(tcp.DestinationAddress); tempPacket.sourcePort = Convert.ToString(tcp.SourcePort); tempPacket.destinationPort = Convert.ToString(tcp.DestinationPort); tempPacket.packetMessage = Convert.ToString(tcp.Data); packetsList.Add(tempPacket); string[] row = { packetsList[countOfPacketCaptures].packetType, packetsList[countOfPacketCaptures].sourceAddress, packetsList[countOfPacketCaptures].destinationAddress, packetsList[countOfPacketCaptures].sourcePort, packetsList[countOfPacketCaptures].destinationPort, packetsList[countOfPacketCaptures].packetMessage }; try { listView1.Items.Add(packetsList[countOfPacketCaptures].packetType, packetsList[countOfPacketCaptures].sourceAddress, packetsList[countOfPacketCaptures].destinationAddress, packetsList[countOfPacketCaptures].sourcePort, packetsList[countOfPacketCaptures].destinationPort, packetsList[countOfPacketCaptures].packetMessage) ; countOfPacketCaptures++; lblCapturesLabels.Text = Convert.ToString(countOfPacketCaptures);} catch (Exception e) { } } else if (packet is UDPPacket) { UDPPacket udp = (UDPPacket)packet; myPacket tempPacket = new myPacket(); tempPacket.packetType = "UDP"; tempPacket.sourceAddress = Convert.ToString(udp.SourceAddress); tempPacket.destinationAddress = Convert.ToString(udp.DestinationAddress); tempPacket.sourcePort = Convert.ToString(udp.SourcePort); tempPacket.destinationPort = Convert.ToString(udp.DestinationPort); tempPacket.packetMessage = Convert.ToString(udp.Data); packetsList.Add(tempPacket); string[] row = { packetsList[countOfPacketCaptures].packetType, packetsList[countOfPacketCaptures].sourceAddress, packetsList[countOfPacketCaptures].destinationAddress, packetsList[countOfPacketCaptures].sourcePort, packetsList[countOfPacketCaptures].destinationPort, packetsList[countOfPacketCaptures].packetMessage }; try { dgwPacketInfo.Rows.Add(row); countOfPacketCaptures++; lblCapturesLabels.Text = Convert.ToString(countOfPacketCaptures); } catch (Exception e) { } } } }

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• LSP packet modify

  - by kellogs

  Hello, anybody care to share some insights on how to use LSP for packet modifying ? I am using the non IFS subtype and I can see how (pseudo?) packets first enter WSPRecv. But how do I modify them ? My inquiry is about one single HTTP response that causes WSPRecv to be called 3 times :((. I need to modify several parts of this response, but since it comes in 3 slices, it is pretty hard to modify it accordingly. And, maybe on other machines or under different conditions (such as high traffic) there would only be one sole WSPRecv call, or maybe 10 calls. What is the best way to work arround this (please no NDIS :D), and how to properly change the buffer (lpBuffers-buf) by increasing it ? int WSPAPI WSPRecv( SOCKET s, LPWSABUF lpBuffers, DWORD dwBufferCount, LPDWORD lpNumberOfBytesRecvd, LPDWORD lpFlags, LPWSAOVERLAPPED lpOverlapped, LPWSAOVERLAPPED_COMPLETION_ROUTINE lpCompletionRoutine, LPWSATHREADID lpThreadId, LPINT lpErrno ) { LPWSAOVERLAPPEDPLUS ProviderOverlapped = NULL; SOCK_INFO *SocketContext = NULL; int ret = SOCKET_ERROR; *lpErrno = NO_ERROR; // // Find our provider socket corresponding to this one // SocketContext = FindAndRefSocketContext(s, lpErrno); if ( NULL == SocketContext ) { dbgprint( "WSPRecv: FindAndRefSocketContext failed!" ); goto cleanup; } // // Check for overlapped I/O // if ( NULL != lpOverlapped ) { /*bla bla .. not interesting in my case*/ } else { ASSERT( SocketContext->Provider->NextProcTable.lpWSPRecv ); SetBlockingProvider(SocketContext->Provider); ret = SocketContext->Provider->NextProcTable.lpWSPRecv( SocketContext->ProviderSocket, lpBuffers, dwBufferCount, lpNumberOfBytesRecvd, lpFlags, lpOverlapped, lpCompletionRoutine, lpThreadId, lpErrno); SetBlockingProvider(NULL); //is this the place to modify packet length and contents ? if (strstr(lpBuffers->buf, "var mapObj = null;")) { int nLen = strlen(lpBuffers->buf) + 200; /*CHAR *szNewBuf = new CHAR[]; CHAR *pIndex; pIndex = strstr(lpBuffers->buf, "var mapObj = null;"); nLen = strlen(strncpy(szNewBuf, lpBuffers->buf, (pIndex - lpBuffers->buf) * sizeof (CHAR))); nLen = strlen(strncpy(szNewBuf + nLen * sizeof(CHAR), "var com = null;\r\n", 17 * sizeof(CHAR))); pIndex += 18 * sizeof(CHAR); nLen = strlen(strncpy(szNewBuf + nLen * sizeof(CHAR), pIndex, 1330 * sizeof (CHAR))); nLen = strlen(strncpy(szNewBuf + nLen * sizeof(CHAR), "if (com == null)\r\n" \ "com = new ActiveXObject(\"InterCommJS.Gateway\");\r\n" \ "com.lat = latitude;\r\n" \ "com.lon = longitude;\r\n}", 111 * sizeof (CHAR))); pIndex = strstr(szNewBuf, "Content-Length:"); pIndex += 16 * sizeof(CHAR); strncpy(pIndex, "1465", 4 * sizeof(CHAR)); lpBuffers->buf = szNewBuf; lpBuffers->len += 128;*/ } if ( SOCKET_ERROR != ret ) { SocketContext->BytesRecv += *lpNumberOfBytesRecvd; } } cleanup: if ( NULL != SocketContext ) DerefSocketContext( SocketContext, lpErrno ); return ret; } Thank you

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• Tools to help with analysing log files

  - by peter

  I am developing a C# .NET application. In the app.config file I add trace logging as shown, <?xml version="1.0" encoding="UTF-8" ?> <configuration> <system.diagnostics> <trace autoflush="true" /> <sources> <source name="System.Net.Sockets" maxdatasize="1024"> <listeners> <add name="MyTraceFile"/> </listeners> </source> </sources> <sharedListeners> <add name="MyTraceFile" type="System.Diagnostics.TextWriterTraceListener" initializeData="System.Net.trace.log" /> </sharedListeners> <switches> <add name="System.Net" value="Verbose" /> </switches> </system.diagnostics> </configuration> Are there any good tools around to analyse the log file that is output? The output looks like this, System.Net.Sockets Verbose: 0 : [5900] Data from Socket#8764489::Send DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000000 : 4D 49 4D 45 2D 56 65 72-73 69 6F 6E 3A 20 31 2E : MIME-Version: 1. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000060 : 65 3A 20 37 20 41 70 72-20 32 30 31 30 20 31 35 : e: 7 Apr 2010 15 DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000070 : 3A 32 32 3A 34 30 20 2B-31 32 30 30 0D 0A 53 75 : :22:40 +1200..Su DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000080 : 62 6A 65 63 74 3A 20 5B-45 72 72 6F 72 5D 20 45 : bject: [Error] E DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000090 : 78 63 65 70 74 69 6F 6E-20 69 6E 20 53 79 6E 63 : xception in Sync DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000A0 : 53 65 72 76 69 63 65 20-28 32 30 30 38 2E 30 2E : Service (2008.0. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000B0 : 33 30 34 2E 31 32 33 34-32 29 0D 0A 43 6F 6E 74 : 304.12342)..Cont DateTime=2010-04-07T03:22:40.1067012Z Is there anything that can take the output shown above (my output is a text file 100mb in size), group together packets, and help out with finding particular issues I would like to hear about it. Thanks.

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• architecture python question

  - by tom smith

  hi. creating a distributed crawling python app. it consists of a master server, and associated client apps that will run on client servers. the purpose of the client app is to run across a targeted site, to extract specific data. the clients need to go "deep" within the site, behind multiple levels of forms, so each client is specifically geared towards a given site. each client app looks something like main: parse initial url call function level1 (data1) function level1 (data) parse the url, for data1 use the required xpath to get the dom elements call the next function call level2 (data) function level2 (data2) parse the url, for data2 use the required xpath to get the dom elements call the next function call level3 function level3 (dat3) parse the url, for data3 use the required xpath to get the dom elements call the next function call level4 function level4 (data) parse the url, for data4 use the required xpath to get the dom elements at the final function.. --all the data output, and eventually returned to the server --at this point the data has elements from each function... my question: given that the number of calls that is made to the child function by the current function varies, i'm trying to figure out the best approach. each function essentialy fetches a page of content, and then parses the page using a number of different XPath expressions, combined with different regex expressions depending on the site/page. if i run a client on a single box, as a sequential process, it'll take awhile, but the load on the box is rather small. i've thought of attempting to implement the child functions as threads from the current function, but that could be a nightmare, as well as quickly bring the "box" to its knees! i've thought of breaking the app up in a manner that would allow the master to essentially pass packets to the client boxes, in a way to allow each client/function to be run directly from the master. this process requires a bit of rewrite, but it has a number of advantages. a bunch of redundancy, and speed. it would detect if a section of the process was crashing and restart from that point. but not sure if it would be any faster... i'm writing the parsing scripts in python.. so... any thoughts/comments would be appreciated... i can get into a great deal more detail, but didn't want to bore anyone!! thanks! tom

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• Tools to Help out with

  - by peter

  I am developing a C# .NET application. In the app.config file I add trace logging as shown, <?xml version="1.0" encoding="UTF-8" ?> <configuration> <system.diagnostics> <trace autoflush="true" /> <sources> <source name="System.Net.Sockets" maxdatasize="1024"> <listeners> <add name="MyTraceFile"/> </listeners> </source> </sources> <sharedListeners> <add name="MyTraceFile" type="System.Diagnostics.TextWriterTraceListener" initializeData="System.Net.trace.log" /> </sharedListeners> <switches> <add name="System.Net" value="Verbose" /> </switches> </system.diagnostics> </configuration> Are there any good tools around to analyse the log file that is output? The output looks like this, System.Net.Sockets Verbose: 0 : [5900] Data from Socket#8764489::Send DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000000 : 4D 49 4D 45 2D 56 65 72-73 69 6F 6E 3A 20 31 2E : MIME-Version: 1. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000060 : 65 3A 20 37 20 41 70 72-20 32 30 31 30 20 31 35 : e: 7 Apr 2010 15 DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000070 : 3A 32 32 3A 34 30 20 2B-31 32 30 30 0D 0A 53 75 : :22:40 +1200..Su DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000080 : 62 6A 65 63 74 3A 20 5B-45 72 72 6F 72 5D 20 45 : bject: [Error] E DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000090 : 78 63 65 70 74 69 6F 6E-20 69 6E 20 53 79 6E 63 : xception in Sync DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000A0 : 53 65 72 76 69 63 65 20-28 32 30 30 38 2E 30 2E : Service (2008.0. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000B0 : 33 30 34 2E 31 32 33 34-32 29 0D 0A 43 6F 6E 74 : 304.12342)..Cont DateTime=2010-04-07T03:22:40.1067012Z Is there anything that can take the output shown above (my output is a text file 100mb in size), group together packets, and help out with finding particular issues I would like to hear about it. Thanks.

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• Anybody Know of any Tools to help Analysing .NET Trace Log Files?

  - by peter

  I am developing a C# .NET application. In the app.config file I add trace logging as shown, <?xml version="1.0" encoding="UTF-8" ?> <configuration> <system.diagnostics> <trace autoflush="true" /> <sources> <source name="System.Net.Sockets" maxdatasize="1024"> <listeners> <add name="MyTraceFile"/> </listeners> </source> </sources> <sharedListeners> <add name="MyTraceFile" type="System.Diagnostics.TextWriterTraceListener" initializeData="System.Net.trace.log" /> </sharedListeners> <switches> <add name="System.Net" value="Verbose" /> </switches> </system.diagnostics> </configuration> Are there any good tools around to analyse the log file that is output? The output looks like this, System.Net.Sockets Verbose: 0 : [5900] Data from Socket#8764489::Send DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000000 : 4D 49 4D 45 2D 56 65 72-73 69 6F 6E 3A 20 31 2E : MIME-Version: 1. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000060 : 65 3A 20 37 20 41 70 72-20 32 30 31 30 20 31 35 : e: 7 Apr 2010 15 DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000070 : 3A 32 32 3A 34 30 20 2B-31 32 30 30 0D 0A 53 75 : :22:40 +1200..Su DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000080 : 62 6A 65 63 74 3A 20 5B-45 72 72 6F 72 5D 20 45 : bject: [Error] E DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000090 : 78 63 65 70 74 69 6F 6E-20 69 6E 20 53 79 6E 63 : xception in Sync DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000A0 : 53 65 72 76 69 63 65 20-28 32 30 30 38 2E 30 2E : Service (2008.0. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000B0 : 33 30 34 2E 31 32 33 34-32 29 0D 0A 43 6F 6E 74 : 304.12342)..Cont DateTime=2010-04-07T03:22:40.1067012Z Is there anything that can take the output shown above (my output is a text file 100mb in size), group together packets, and help out with finding particular issues I would like to hear about it. Thanks.

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• Python and C++ Sockets converting packet data

  - by yeus

  First of all, to clarify my goal: There exist two programs written in C in our laboratory. I am working on a Proxy Server (bidirectional) for them (which will also mainpulate the data). And I want to write that proxy server in Python. It is important to know that I know close to nothing about these two programs, I only know the definition file of the packets. Now: assuming a packet definition in one of the C++ programs reads like this: unsigned char Packet[0x32]; // Packet[Length] int z=0; Packet[0]=0x00; // Spare Packet[1]=0x32; // Length Packet[2]=0x01; // Source Packet[3]=0x02; // Destination Packet[4]=0x01; // ID Packet[5]=0x00; // Spare for(z=0;z<=24;z+=8) { Packet[9-z/8]=((int)(720000+armcontrolpacket->dof0_rot*1000)/(int)pow((double)2,(double)z)); Packet[13-z/8]=((int)(720000+armcontrolpacket->dof0_speed*1000)/(int)pow((double)2,(double)z)); Packet[17-z/8]=((int)(720000+armcontrolpacket->dof1_rot*1000)/(int)pow((double)2,(double)z)); Packet[21-z/8]=((int)(720000+armcontrolpacket->dof1_speed*1000)/(int)pow((double)2,(double)z)); Packet[25-z/8]=((int)(720000+armcontrolpacket->dof2_rot*1000)/(int)pow((double)2,(double)z)); Packet[29-z/8]=((int)(720000+armcontrolpacket->dof2_speed*1000)/(int)pow((double)2,(double)z)); Packet[33-z/8]=((int)(720000+armcontrolpacket->dof3_rot*1000)/(int)pow((double)2,(double)z)); Packet[37-z/8]=((int)(720000+armcontrolpacket->dof3_speed*1000)/(int)pow((double)2,(double)z)); Packet[41-z/8]=((int)(720000+armcontrolpacket->dof4_rot*1000)/(int)pow((double)2,(double)z)); Packet[45-z/8]=((int)(720000+armcontrolpacket->dof4_speed*1000)/(int)pow((double)2,(double)z)); Packet[49-z/8]=((int)armcontrolpacket->timestamp/(int)pow(2.0,(double)z)); } if(SendPacket(sock,(char*)&Packet,sizeof(Packet))) return 1; return 0; What would be the easiest way to receive that data, convert it into a readable python format, manipulate them and send them forward to the receiver?

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• How to stream semi-live audio over internet

  - by Thomas Tempelmann

  I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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• IEnumerable<T> ToArray usage, is it a copy or a pointer?

  - by Daniel

  I am parsing an arbitrary length byte array that is going to be passed around to a few different layers of parsing. Each parser creates a Header and a Packet payload just like any ordinary encapsulation. And my problem lies in how the encapsulation holds its packet byte array payload. Say i have a 100 byte array, and it has 3 levels of encapsulation. 3 packet objects will be created and i want to set the payload of these packets to the corresponding position in the byte array of the packet. For example lets say the payload size is 20 for all levels, then imagine it has a public byte[] Payload on each object. However the problem is that this byte[] Payload is a copy of the original 100 bytes. So i'm going to end up with 160 bytes in memory instead of 100. If it were in c++ i could just easily use a pointer however i'm writing this in c#. So i created the following class: public class PayloadSegment<T> : IEnumerable<T> { public readonly T[] Array; public readonly int Offset; public readonly int Count; public PayloadSegment(T[] array, int offset, int count) { this.Array = array; this.Offset = offset; this.Count = count; } public T this[int index] { get { if (index < 0 || index >= this.Count) throw new IndexOutOfRangeException(); else return Array[Offset + index]; } set { if (index < 0 || index >= this.Count) throw new IndexOutOfRangeException(); else Array[Offset + index] = value; } } public IEnumerator<T> GetEnumerator() { for (int i = Offset; i < Offset + Count; i++) yield return Array[i]; } System.Collections.IEnumerator System.Collections.IEnumerable.GetEnumerator() { IEnumerator<T> enumerator = this.GetEnumerator(); while (enumerator.MoveNext()) { yield return enumerator.Current; } } } This way i can simply reference a position inside the original byte array but use positional indexing. However if i do something like: PayloadSegment<byte> something = new PayloadSegment<byte>(someArray, 5, 10); byte[] somethingArray = something.ToArray(); Will the somethingArray be a copy of the bytes, or a reference to the original PayloadSegment which in turn is a reference to the original byte array? Sorry it was hard to word this lol _<

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• Monitoring UDP socket in glib(mm) eats up CPU time

  - by Gyorgy Szekely

  Hi, I have a GTKmm Windows application (built with MinGW) that receives UDP packets (no sending). The socket is native winsock and I use glibmm IOChannel to connect it to the application main loop. The socket is read with recvfrom. My problem is: this setup eats 25% percent CPU time on a 3GHz workstation. Can somebody tell me why? The application is idle in this case, and if I remove the UDP code, CPU usage drops down to almost zero. As the application has to perform some CPU intensive tasks, I could image better ways to spend that 25% Here are some code excerpts: (sorry for the printf's ;) ) /* bind */ void UDPInterface::bindToPort(unsigned short port) { struct sockaddr_in target; WSADATA wsaData; target.sin_family = AF_INET; target.sin_port = htons(port); target.sin_addr.s_addr = 0; if ( WSAStartup ( 0x0202, &wsaData ) ) { printf("WSAStartup failed!\n"); exit(0); // :) WSACleanup(); } sock = socket( AF_INET, SOCK_DGRAM, 0 ); if (sock == INVALID_SOCKET) { printf("invalid socket!\n"); exit(0); } if (bind(sock,(struct sockaddr*) &target, sizeof(struct sockaddr_in) ) == SOCKET_ERROR) { printf("failed to bind to port!\n"); exit(0); } printf("[UDPInterface::bindToPort] listening on port %i\n", port); } /* read */ bool UDPInterface::UDPEvent(Glib::IOCondition io_condition) { recvfrom(sock, (char*)buf, BUF_SIZE*4, 0, NULL, NULL); /* process packet... */ } /* glibmm connect */ Glib::RefPtr channel = Glib::IOChannel::create_from_win32_socket(udp.sock); Glib::signal_io().connect( sigc::mem_fun(udp, &UDPInterface::UDPEvent), channel, Glib::IO_IN ); I've read here in some other question, and also in glib docs (g_io_channel_win32_new_socket()) that the socket is put into nonblocking mode, and it's "a side-effect of the implementation and unavoidable". Does this explain the CPU effect, it's not clear to me? Whether or not I use glib to access the socket or call recvfrom() directly doesn't seem to make much difference, since CPU is used up before any packet arrives and the read handler gets invoked. Also glibmm docs state that it's ok to call recvfrom() even if the socket is polled (Glib::IOChannel::create_from_win32_socket()) I've tried compiling the program with -pg and created a per function cpu usage report with gprof. This wasn't usefull because the time is not spent in my program, but in some external glib/glibmm dll.

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• Correct usage of socket_select().

  - by Mark Tomlin

  What is the correct way to use socket_select within PHP to send and receive data? I have a connection to the server that allows for both TCP & UDP packet connections, I am utilizing both. Within these connections I'm both sending and receiving packets on the same port, but the TCP packet will be sent on one port (29999) and UDP will be sent on another port (30000). The transmission type will be that of AF_INET. The IP address will be loopback 127.0.0.1. I have many questions on how to create a socket connection within this scenario. For example, is it better to use socket_create_pair to make the connection, or use just socket_create followed by socket_connect, and then implement socket_select? There is a chance that no data will be sent from the server to the client, and it is up to the client to maintain the connection. This will be done by utilizing the time out function within the socket_select call. Should no data be sent within the time limit, the socket_select function will break and a keep alive packet can then be sent. The following script is of the client. // Create $TCP = socket_create(AF_INET, SOCK_STREAM, SOL_TCP); $UDP = socket_create(AF_INET, SOCK_DGRAM, SOL_UDP); // Misc $isAlive = TRUE; $UDPPort = 30000; define('ISP_ISI', 1); // Connect socket_connect($TCP, '127.0.0.1', 29999); socket_connect($UDP, '127.0.0.1', $UDPPort); // Construct Parameters $recv = array($TCP, $UDP); $null = NULL; // Make The Packet to Send. $packet = pack('CCCxSSxCSa16a16', 44, ISP_ISI, 1, $UDPPort, 0, '!', 0, 'AdminPass', 'SocketSelect'); // Send ISI (InSim Init) Packet socket_write($TCP, $packet); /* Main Program Loop */ while ($isAlive == TRUE) { // Socket Select $sock = socket_select($recv, $null, $null, 5); // Check Status if ($sock === FALSE) $isAlive = FALSE; # Error else if ($sock > 0) # How does one check to find what socket changed? else # Something else happed, don't know what as it's not in the documentation, Could this be our timeout getting tripped? }

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• Displaying a notification when bluetooth is disconnected - Android

  - by Ryan T

  I am trying to create a program that will display a notification to the user if a Blue tooth device suddenly comes out of range from my Android device. I currently have the following code but no notification is displayed. I was wondering if it was possible I shouldn't use ACTION_ACL_DISCONNECTED because I believe the bluetooth stack would be expecting packets that state a disconnect is requested. My requirements state that the bluetooth device will disconnect without warning. Thank you for any assistance! BluetoothNotification.java: //This is where the notification is created. import android.app.Activity; import android.app.Notification; import android.app.NotificationManager; import android.app.PendingIntent; import android.content.Context; import android.content.Intent; import android.os.Bundle; import android.app.Activity; import android.app.Notification; import android.app.NotificationManager; import android.app.PendingIntent; import android.content.Context; import android.content.Intent; import android.os.Bundle; public class BluetoothNotification extends Activity { public static final int NOTIFICATION_ID = 1; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); /** Define configuration for our notification */ int icon = R.drawable.logo; CharSequence tickerText = "This is a sample notification"; long when = System.currentTimeMillis(); Context context = getApplicationContext(); CharSequence contentTitle = "Sample notification"; CharSequence contentText = "This notification has been generated as a result of BT Disconnecting"; Intent notificationIntent = new Intent(this, BluetoothNotification.class); PendingIntent contentIntent = PendingIntent.getActivity(this, 0, notificationIntent, 0); /** Initialize the Notification using the above configuration */ final Notification notification = new Notification(icon, tickerText, when); notification.setLatestEventInfo(context, contentTitle, contentText, contentIntent); /** Retrieve reference from NotificationManager */ String ns = Context.NOTIFICATION_SERVICE; final NotificationManager mNotificationManager = (NotificationManager) getSystemService(ns); mNotificationManager.notify(NOTIFICATION_ID, notification); finish(); } } Snippet from OnCreate: //Located in Controls.java IntentFilter filter1 = new IntentFilter(BluetoothDevice.ACTION_ACL_DISCONNECTED); this.registerReceiver(mReceiver, filter1); Snippet from Controls.java: private final BroadcastReceiver mReceiver = new BroadcastReceiver() { @Override public void onReceive(Context context, Intent intent) { String action = intent.getAction(); BluetoothDevice device = intent.getParcelableExtra(BluetoothDevice.EXTRA_DEVICE); if (BluetoothDevice.ACTION_ACL_DISCONNECTED.equals(action)) { //Device has disconnected NotificationManager nm = (NotificationManager) getSystemService(NOTIFICATION_SERVICE); } } };

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• Abnormally disconnected TCP sockets and write timeout

  - by James

  Hello I will try to explain the problem in shortest possible words. I am using c++ builder 2010. I am using TIdTCPServer and sending voice packets to a list of connected clients. Everything works ok untill any client is disconnected abnormally, For example power failure etc. I can reproduce similar disconnect by cutting the ethernet connection of a connected client. So now we have a disconnected socket but as you know it is not yet detected at server side so server will continue to try to send data to that client too. But when server try to write data to that disconnected client ...... Write() or WriteLn() HANGS there in trying to write, It is like it is wating for somekind of Write timeout. This hangs the hole packet distribution process as a result creating a lag in data transmission to all other clients. After few seconds "Socket Connection Closed" Exception is raised and data flow continues. Here is the code try { EnterCriticalSection(&SlotListenersCriticalSection); for(int i=0;i<SlotListeners->Count;i++) { try { //Here the process will HANG for several seconds on a disconnected socket ((TIdContext*) SlotListeners->Objects[i])->Connection->IOHandler->WriteLn("Some DATA"); }catch(Exception &e) { SlotListeners->Delete(i); } } }__finally { LeaveCriticalSection(&SlotListenersCriticalSection); } Ok i already have a keep alive mechanism which disconnect the socket after n seconds of inactivity. But as you can imagine, still this mechnism cant sync exactly with this braodcasting loop because this braodcasting loop is running almost all the time. So is there any Write timeouts i can specify may be through iohandler or something ? I have seen many many threads about "Detecting disconnected tcp socket" but my problem is little different, i need to avoid that hangup for few seconds during the write attempt. So is there any solution ? Or should i consider using some different mechanism for such data broadcasting for example the broadcasting loop put the data packet in some kind of FIFO buffer and client threads continuously check for available data and pick and deliver it to themselves ? This way if one thread hangs it will not stop/delay the over all distribution thread. Any ideas please ? Thanks for your time and help. Regards Jams

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• waiting for 2 different events in a single thread

  - by João Portela

  component A (in C++) - is blocked waiting for alarm signals (not relevant) and IO signals (1 udp socket). has one handler for each of these. component B (java) - has to receive the same information the component A udp socket receives. periodicaly gives instructions that should be sent through component A udp socket. How to join both components? it is strongly desirable that: the changes to attach component B to component A are minimal (its not my code and it is not very pleasent to mess with). the time taken by the new operations (usually communicating with component B) interfere very little with the usual processing time of component A - this means that if the operations are going to take a "some" time I would rather use a thread or something to do them. note: since component A receives udp packets more frequently that it has component B instructions to forward, if necessary, it can only forward the instructions (when available) from the IO handler. my initial ideia was to develop a component C (in C++) that would sit inside the component A code (is this called an adapter?) that when instanciated starts the java process and makes the necessary connections (that not so little overhead in the initialization is not a problem). It would have 2 stacks, one for the data to give component B (lets call it Bstack) and for the data to give component A (lets call it Astack). It would sit on its thread (lets call it new-thread) waiting for data to be available in Bstack to send it over udp, and listen on the udp socket to put data on the Astack. This means that the changes to component A are only: when it receives a new UDP packet put it on the Bstack, and if there is something on the Astack sent it over its UDP socket (I decided for this because this socket would only be used in the main thread). One of the problems is that I don't know how to wait for both of these events at the same time using only one thread. so my questions are: Do I really need to use the main thread to send the data over component A socket or can I do it from the new-thread? (I think the answer is no, but I'm not sure about race conditions on sockets) how to I wait for both events? boost::condition_variable or something similar seems the solution in the case of the stack and boost::asio::io_service io_service.run() seems like the thing to use for the socket. Is there any other alternative solution for this problem that I'm not aware of? Thanks for reading this long text but I really wanted you to understand the problem.

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• How do you unit test the real world?

  - by Kim Sun-wu

  I'm primarily a C++ coder, and thus far, have managed without really writing tests for all of my code. I've decided this is a Bad Idea(tm), after adding new features that subtly broke old features, or, depending on how you wish to look at it, introduced some new "features" of their own. But, unit testing seems to be an extremely brittle mechanism. You can test for something in "perfect" conditions, but you don't get to see how your code performs when stuff breaks. A for instance is a crawler, let's say it crawls a few specific sites, for data X. Do you simply save sample pages, test against those, and hope that the sites never change? This would work fine as regression tests, but, what sort of tests would you write to constantly check those sites live and let you know when the application isn't doing it's job because the site changed something, that now causes your application to crash? Wouldn't you want your test suite to monitor the intent of the code? The above example is a bit contrived, and something I haven't run into (in case you haven't guessed). Let me pick something I have, though. How do you test an application will do its job in the face of a degraded network stack? That is, say you have a moderate amount of packet loss, for one reason or the other, and you have a function DoSomethingOverTheNetwork() which is supposed to degrade gracefully when the stack isn't performing as it's supposed to; but does it? The developer tests it personally by purposely setting up a gateway that drops packets to simulate a bad network when he first writes it. A few months later, someone checks in some code that modifies something subtly, so the degradation isn't detected in time, or, the application doesn't even recognize the degradation, this is never caught, because you can't run real world tests like this using unit tests, can you? Further, how about file corruption? Let's say you're storing a list of servers in a file, and the checksum looks okay, but the data isn't really. You want the code to handle that, you write some code that you think does that. How do you test that it does exactly that for the life of the application? Can you? Hence, brittleness. Unit tests seem to test the code only in perfect conditions(and this is promoted, with mock objects and such), not what they'll face in the wild. Don't get me wrong, I think unit tests are great, but a test suite composed only of them seems to be a smart way to introduce subtle bugs in your code while feeling overconfident about it's reliability. How do I address the above situations? If unit tests aren't the answer, what is? Thanks!

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• How to convert m4a file to aac adts file in Xcode?

  - by Bird Hsuie

  I have a mp4 file copied from iPod lib and saved to my Document for my next step, I need it to convert to .mp3 or .aac(ADTS type) I use this code and failed... -(IBAction)compressFile:(id)sender{ NSLog (@"handleConvertToPCMTapped"); // open an ExtAudioFile NSLog (@"opening %@", exportURL); ExtAudioFileRef inputFile; CheckResult (ExtAudioFileOpenURL((__bridge CFURLRef)exportURL, &inputFile), "ExtAudioFileOpenURL failed"); // prepare to convert to a plain ol' PCM format AudioStreamBasicDescription myPCMFormat; myPCMFormat.mSampleRate = 44100; // todo: or use source rate? myPCMFormat.mFormatID = kAudioFormatMPEGLayer3 ; myPCMFormat.mFormatFlags = kAudioFormatFlagsCanonical; myPCMFormat.mChannelsPerFrame = 2; myPCMFormat.mFramesPerPacket = 1; myPCMFormat.mBitsPerChannel = 16; myPCMFormat.mBytesPerPacket = 4; myPCMFormat.mBytesPerFrame = 4; CheckResult (ExtAudioFileSetProperty(inputFile, kExtAudioFileProperty_ClientDataFormat, sizeof (myPCMFormat), &myPCMFormat), "ExtAudioFileSetProperty failed"); // allocate a big buffer. size can be arbitrary for ExtAudioFile. // you have 64 KB to spare, right? UInt32 outputBufferSize = 0x10000; void* ioBuf = malloc (outputBufferSize); UInt32 sizePerPacket = myPCMFormat.mBytesPerPacket; UInt32 packetsPerBuffer = outputBufferSize / sizePerPacket; // set up output file NSString *outputPath = [myDocumentsDirectory() stringByAppendingPathComponent:@"m_export.mp3"]; NSURL *outputURL = [NSURL fileURLWithPath:outputPath]; NSLog (@"creating output file %@", outputURL); AudioFileID outputFile; CheckResult(AudioFileCreateWithURL((__bridge CFURLRef)outputURL, kAudioFileCAFType, &myPCMFormat, kAudioFileFlags_EraseFile, &outputFile), "AudioFileCreateWithURL failed"); // start convertin' UInt32 outputFilePacketPosition = 0; //in bytes while (true) { // wrap the destination buffer in an AudioBufferList AudioBufferList convertedData; convertedData.mNumberBuffers = 1; convertedData.mBuffers[0].mNumberChannels = myPCMFormat.mChannelsPerFrame; convertedData.mBuffers[0].mDataByteSize = outputBufferSize; convertedData.mBuffers[0].mData = ioBuf; UInt32 frameCount = packetsPerBuffer; // read from the extaudiofile CheckResult (ExtAudioFileRead(inputFile, &frameCount, &convertedData), "Couldn't read from input file"); if (frameCount == 0) { printf ("done reading from file"); break; } // write the converted data to the output file CheckResult (AudioFileWritePackets(outputFile, false, frameCount, NULL, outputFilePacketPosition / myPCMFormat.mBytesPerPacket, &frameCount, convertedData.mBuffers[0].mData), "Couldn't write packets to file"); NSLog (@"Converted %ld bytes", outputFilePacketPosition); // advance the output file write location outputFilePacketPosition += (frameCount * myPCMFormat.mBytesPerPacket); } // clean up ExtAudioFileDispose(inputFile); AudioFileClose(outputFile); // show size in label NSLog (@"checking file at %@", outputPath); [self transMitFile:outputPath]; if ([[NSFileManager defaultManager] fileExistsAtPath:outputPath]) { NSError *fileManagerError = nil; unsigned long long fileSize = [[[NSFileManager defaultManager] attributesOfItemAtPath:outputPath error:&fileManagerError] fileSize]; } any suggestion?.......thanks for your great help!

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