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• Problems Installing slapd On Ubuntu Server 11.10

  - by Zach Dziura

  I know that there's a Ubuntu-specific StackExchange website, but I thought that I'd ask here because it's a server-specific question. If I'm wrong in my logic... Well, you people are better at this than I am! O=) On with the show! I'm in the process of installing Oracle Database 11g R2 Standard Edition onto Ubuntu Server 11.10. I found a guide on the Oracle Support Forums that walks you through the process fairly easily. Unfortunately, I'm running into issues installing one particular dependency: slapd. When I go to install it, I get this error message: (Reading database ... 64726 files and directories currently installed.) Unpacking slapd (from .../slapd_2.4.25-1.1ubuntu4.1_amd64.deb) ... Processing triggers for man-db ... Processing triggers for ufw ... Processing triggers for ureadahead ... Setting up slapd (2.4.25-1.1ubuntu4.1) ... Usage: slappasswd [options] -c format crypt(3) salt format -g generate random password -h hash password scheme -n omit trailing newline -s secret new password -u generate RFC2307 values (default) -v increase verbosity -T file read file for new password Creating initial configuration... Loading the initial configuration from the ldif file () failed with the following error while running slapadd: str2entry: invalid value for attributeType olcRootPW #0 (syntax 1.3.6.1.4.1.1466.115.121.1.15) slapadd: could not parse entry (line=1051) dpkg: error processing slapd (--configure): subprocess installed post-installation script returned error exit status 1 Errors were encountered while processing: slapd E: Sub-process /usr/bin/dpkg returned an error code (1) After much Google searches and forum trolling, I have yet to find a definitive answer as to what's going wrong. The error messages seem straight forward enough, but I have no idea how to debug this. Can anyone offer some assistance? Again, if I'm asking in the wrong place, I apologize. If I'm indeed asking properly, then thank you for any and all help!

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• How to sync client and server at the first frame

  - by wheelinlight

  I'm making a game where an authoritative server sends information to all clients about states and positions for objects in a 3d world. The player can control his character by clicking on the screen to set a destination for the character, much like in the Diablo series. I've read most information I can find online about interpolation, reconciliation, and general networking architecture (Valve's for instance). I think I understand everything but one thing seems to be missing in every article I read. Let say we have an interpolation delay of 100ms, server tickrate=50ms, latency=200ms; How do I know when 100ms has past on the client? If the server sends the first update on t=0, can I assume it arrives at t=200, therefore assuming that all packets takes the same amount of time to reach the client? What if the first packet arrives a little quick, for instance at t=150. I would then be starting the client with t=150 and at t=250 it will think it has past 100ms since its connect to the server when it in fact only 50ms has past. Hopefully the above paragraph is understandable. The summarized question would be: How do I know at what tick to start simulating the client? EDIT: This is how I ended up doing it: The client keeps a clock (approximately) in sync with the server. The client then simulates the world at simulationTime = syncedTime - avg(RTT)/2 - interpolationTime The round-trip time can fluctuate so therefore I average it out over time. By only keeping the most recent values when calculating the average I hope to adapt to more permanent changes in latency. It's still to early to draw any conclusion. I'm currently simulating bad network connections, but it's looking good so far. Anyone see any possible problems?

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• Computer becomes unreachable on lan after some time

  - by Ashfame

  I work on my laptop and ssh into my desktop. I use a lot of key based authentication for many servers for work but recently I couldn't login because ssh would pick up and try all the keys and it stops trying before ultimately falling back to password based login. So right now I am using this command: ssh -X -o PubkeyAuthentication=no [email protected] #deskto The issue is after sometime the desktop would just become unreachable from laptop. I won't be able to open its localhost through IP and today I tried ping'in it and found a weird thing. Instead of 192.168.1.4, it tries to ping 192.168.1.3 which I am sure is the root cause as it just can't reach 192.168.1.4 when its actually trying for 192.168.1.3 Ping command output: ashfame@ashfame-xps:~$ ping 192.168.1.4 PING 192.168.1.4 (192.168.1.4) 56(84) bytes of data. From 192.168.1.3 icmp_seq=1 Destination Host Unreachable From 192.168.1.3 icmp_seq=2 Destination Host Unreachable From 192.168.1.3 icmp_seq=3 Destination Host Unreachable From 192.168.1.3 icmp_seq=4 Destination Host Unreachable From 192.168.1.3 icmp_seq=5 Destination Host Unreachable From 192.168.1.3 icmp_seq=6 Destination Host Unreachable From 192.168.1.3 icmp_seq=7 Destination Host Unreachable From 192.168.1.3 icmp_seq=8 Destination Host Unreachable From 192.168.1.3 icmp_seq=9 Destination Host Unreachable ^C --- 192.168.1.4 ping statistics --- 10 packets transmitted, 0 received, +9 errors, 100% packet loss, time 9047ms pipe 3 Also the ping command message comes in multiple and not one by one. (izx answer's the weirdness I thought there was in ping command.) I did check for desktop, its local IP is still the same, so something is going on in my laptop. Any ideas? P.S. - Laptop runs Ubuntu 12.04 & Desktop runs Ubuntu 11.10 Laptop is connected through wifi to router and Desktop is connected through LAN to router. Update: Even after setting up static IP leases in router settings, I again ran into this issue.

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• Design help with parallel process

  - by brazc0re

  I am re-factoring some code and an having an issue with retrieving data from two parallel processes. I have an application that sends packets back and forth via different mediums (ex: RS232, TCP/IP, etc). The jist if of this question is that there are two parallel processes going on. I hope the picture below displays what is going on better than I can word it: SetupRS232() class creates a new instance of the SerialPort by: SerialPort serialPort = new SerialPort(); My question is, what is the best way that the Communicator() class, which sends out the packet via the respective medium, get access to the SerialPort object from the SetupRS232 class? I can do it with a Singleton but have heard that they are generally not the best design to go by. I am trying to follow SRP but I do feel like I am doing something wrong here. Communicator() will need to go out of it's way to get access to SetupRS232() to get access to the SerialPort class. I actually haven't found a way to even get access to it. Would designing each medium class, for example, SetupRS232(), SetupTCPIP, as a singleton be the best way to approach this problem?

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• NMap 6.01

  - by TATWORTH

  NMap 6.01 has been released at http://nmap.org/download.html"Nmap ("Network Mapper") is a free and open source (license) utility for network discovery and security auditing. Many systems and network administrators also find it useful for tasks such as network inventory, managing service upgrade schedules, and monitoring host or service uptime. Nmap uses raw IP packets in novel ways to determine what hosts are available on the network, what services (application name and version) those hosts are offering, what operating systems (and OS versions) they are running, what type of packet filters/firewalls are in use, and dozens of other characteristics. It was designed to rapidly scan large networks, but works fine against single hosts. Nmap runs on all major computer operating systems, and official binary packages are available for Linux, Windows, and Mac OS X. In addition to the classic command-line Nmap executable, the Nmap suite includes an advanced GUI and results viewer (Zenmap), a flexible data transfer, redirection, and debugging tool (Ncat), a utility for comparing scan results (Ndiff), and a packet generation and response analysis tool (Nping)."Home page is at http://nmap.org/  Nmap is free to download and use. You can download the source and compile it yourself if you so require.

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• Strange traffic on fresh Ubuntu Server install

  - by Fishy

  I've just installed Ubuntu Server on my home box after becoming partially familiar with it at work and wanting to train up as a Pen Tester. I installed the latest version on a logical partition (the main one contained Win7), and selected none of the extra modules (I think). I installed ngrep and fired it up (along with TCPdump) and immediately saw some strange traffic which I am unable to identify. My pc is sending out UDP packets every couple of seconds to a seemingly random series of IP addresses, all on the same port (47669 - though I did also see it use another port for a while). I watched it do this for about 20 mins, whilst trying to work out why it was doing it. The only other traffic was the odd ARP request for the router and SSDP UPnP broadcasts from the router. Anyone know what this is, or have any advice on how best to find out? Thanks. EDIT: Actually, it's not my box generating the traffic. It's receiving the traffic on that port, from a series of IP addresses, and returning 'port unreachable' messages.

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• networking without port forwarding

  - by Wallacoloo

  I'm trying to add networking functionality to my game. I want any user to be able to host the game, and anyone to be able to connect as a client. The client sends info to the host about their player's position, etc. When the host receives a message, it validates it and then broadcasts it to its other clients. I will primarily be dealing with UDP, but will also need TCP for chat & lobby stuff. The problem is that I can't seem to get a packet sent from the client to the host or the other way around without enabling port forwarding on my router. But I don't think this is necessary. I believe the reason I need port forwarding is because I want to send a packet from 1 computer on a LAN to another computer on a different LAN, but neither of them have a global ip address since they're in a LAN. So really, I can only send packets targeting the other network's router, which must forward it on to the machine I want to reach. So how can I do this without port forwarding? Somehow a web server can communicate with my computer, which doesn't have a global ip, without port forwarding. And I've played plenty of multi-player games that don't require me to enable port forwarding. So it must be possible. Btw, I'm using SDL_Net. I don't think this will change anything though.

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• Cannot determine ethernet address for proxy ARP on PPTP

  - by Linux Intel

  I installed pptp server on a centos 6 64bit server PPTP Server ip : 55.66.77.10 PPTP Local ip : 10.0.0.1 Client1 IP : 10.0.0.60 centos 5 64bit Client2 IP : 10.0.0.61 centos5 64bit PPTP Server can ping Client1 And client 1 can ping PPTP Server PPTP Server can ping Client2 And client 2 can ping PPTP Server The problem is client 1 can not ping Client 2 and i get this error also on PPTP server error log Cannot determine ethernet address for proxy ARP Ping from Client2 to Client1 PING 10.0.0.60 (10.0.0.60) 56(84) bytes of data. --- 10.0.0.60 ping statistics --- 6 packets transmitted, 0 received, 100% packet loss, time 5000ms route -n on PPTP Server Destination Gateway Genmask Flags Metric Ref Use Iface 10.0.0.60 0.0.0.0 255.255.255.255 UH 0 0 0 ppp0 10.0.0.61 0.0.0.0 255.255.255.255 UH 0 0 0 ppp1 55.66.77.10 0.0.0.0 255.255.255.248 U 0 0 0 eth0 10.0.0.0 0.0.0.0 255.0.0.0 U 0 0 0 eth0 0.0.0.0 55.66.77.19 0.0.0.0 UG 0 0 0 eth0 route -n On Client 1 Destination Gateway Genmask Flags Metric Ref Use Iface 10.0.0.1 0.0.0.0 255.255.255.255 UH 0 0 0 ppp0 55.66.77.10 70.14.13.19 255.255.255.255 UGH 0 0 0 eth0 10.0.0.0 0.0.0.0 255.0.0.0 U 0 0 0 eth1 0.0.0.0 70.14.13.19 0.0.0.0 UG 0 0 0 eth0 route -n On Client 2 Destination Gateway Genmask Flags Metric Ref Use Iface 10.0.0.1 0.0.0.0 255.255.255.255 UH 0 0 0 ppp0 55.66.77.10 84.56.120.60 255.255.255.255 UGH 0 0 0 eth1 10.0.0.0 0.0.0.0 255.0.0.0 U 0 0 0 eth0 0.0.0.0 84.56.120.60 0.0.0.0 UG 0 0 0 eth1 cat /etc/ppp/options.pptpd on PPTP server ############################################################################### # $Id: options.pptpd,v 1.11 2005/12/29 01:21:09 quozl Exp $ # # Sample Poptop PPP options file /etc/ppp/options.pptpd # Options used by PPP when a connection arrives from a client. # This file is pointed to by /etc/pptpd.conf option keyword. # Changes are effective on the next connection. See "man pppd". # # You are expected to change this file to suit your system. As # packaged, it requires PPP 2.4.2 and the kernel MPPE module. ############################################################################### # Authentication # Name of the local system for authentication purposes # (must match the second field in /etc/ppp/chap-secrets entries) name pptpd # Strip the domain prefix from the username before authentication. # (applies if you use pppd with chapms-strip-domain patch) #chapms-strip-domain # Encryption # (There have been multiple versions of PPP with encryption support, # choose with of the following sections you will use.) # BSD licensed ppp-2.4.2 upstream with MPPE only, kernel module ppp_mppe.o # {{{ refuse-pap refuse-chap refuse-mschap # Require the peer to authenticate itself using MS-CHAPv2 [Microsoft # Challenge Handshake Authentication Protocol, Version 2] authentication. require-mschap-v2 # Require MPPE 128-bit encryption # (note that MPPE requires the use of MSCHAP-V2 during authentication) require-mppe-128 # }}} # OpenSSL licensed ppp-2.4.1 fork with MPPE only, kernel module mppe.o # {{{ #-chap #-chapms # Require the peer to authenticate itself using MS-CHAPv2 [Microsoft # Challenge Handshake Authentication Protocol, Version 2] authentication. #+chapms-v2 # Require MPPE encryption # (note that MPPE requires the use of MSCHAP-V2 during authentication) #mppe-40 # enable either 40-bit or 128-bit, not both #mppe-128 #mppe-stateless # }}} # Network and Routing # If pppd is acting as a server for Microsoft Windows clients, this # option allows pppd to supply one or two DNS (Domain Name Server) # addresses to the clients. The first instance of this option # specifies the primary DNS address; the second instance (if given) # specifies the secondary DNS address. #ms-dns 10.0.0.1 #ms-dns 10.0.0.2 # If pppd is acting as a server for Microsoft Windows or "Samba" # clients, this option allows pppd to supply one or two WINS (Windows # Internet Name Services) server addresses to the clients. The first # instance of this option specifies the primary WINS address; the # second instance (if given) specifies the secondary WINS address. #ms-wins 10.0.0.3 #ms-wins 10.0.0.4 # Add an entry to this system's ARP [Address Resolution Protocol] # table with the IP address of the peer and the Ethernet address of this # system. This will have the effect of making the peer appear to other # systems to be on the local ethernet. # (you do not need this if your PPTP server is responsible for routing # packets to the clients -- James Cameron) proxyarp # Normally pptpd passes the IP address to pppd, but if pptpd has been # given the delegate option in pptpd.conf or the --delegate command line # option, then pppd will use chap-secrets or radius to allocate the # client IP address. The default local IP address used at the server # end is often the same as the address of the server. To override this, # specify the local IP address here. # (you must not use this unless you have used the delegate option) #10.8.0.100 # Logging # Enable connection debugging facilities. # (see your syslog configuration for where pppd sends to) debug # Print out all the option values which have been set. # (often requested by mailing list to verify options) #dump # Miscellaneous # Create a UUCP-style lock file for the pseudo-tty to ensure exclusive # access. lock # Disable BSD-Compress compression nobsdcomp # Disable Van Jacobson compression # (needed on some networks with Windows 9x/ME/XP clients, see posting to # poptop-server on 14th April 2005 by Pawel Pokrywka and followups, # http://marc.theaimsgroup.com/?t=111343175400006&r=1&w=2 ) novj novjccomp # turn off logging to stderr, since this may be redirected to pptpd, # which may trigger a loopback nologfd # put plugins here # (putting them higher up may cause them to sent messages to the pty) cat /etc/ppp/options.pptp on Client1 and Client2 ############################################################################### # $Id: options.pptp,v 1.3 2006/03/26 23:11:05 quozl Exp $ # # Sample PPTP PPP options file /etc/ppp/options.pptp # Options used by PPP when a connection is made by a PPTP client. # This file can be referred to by an /etc/ppp/peers file for the tunnel. # Changes are effective on the next connection. See "man pppd". # # You are expected to change this file to suit your system. As # packaged, it requires PPP 2.4.2 or later from http://ppp.samba.org/ # and the kernel MPPE module available from the CVS repository also on # http://ppp.samba.org/, which is packaged for DKMS as kernel_ppp_mppe. ############################################################################### # Lock the port lock # Authentication # We don't need the tunnel server to authenticate itself noauth # We won't do PAP, EAP, CHAP, or MSCHAP, but we will accept MSCHAP-V2 # (you may need to remove these refusals if the server is not using MPPE) refuse-pap refuse-eap refuse-chap refuse-mschap # Compression # Turn off compression protocols we know won't be used nobsdcomp nodeflate # Encryption # (There have been multiple versions of PPP with encryption support, # choose which of the following sections you will use. Note that MPPE # requires the use of MSCHAP-V2 during authentication) # # Note that using PPTP with MPPE and MSCHAP-V2 should be considered # insecure: # http://marc.info/?l=pptpclient-devel&m=134372640219039&w=2 # https://github.com/moxie0/chapcrack/blob/master/README.md # http://technet.microsoft.com/en-us/security/advisory/2743314 # http://ppp.samba.org/ the PPP project version of PPP by Paul Mackarras # ppp-2.4.2 or later with MPPE only, kernel module ppp_mppe.o # If the kernel is booted in FIPS mode (fips=1), the ppp_mppe.ko module # is not allowed and PPTP-MPPE is not available. # {{{ # Require MPPE 128-bit encryption #require-mppe-128 # }}} # http://mppe-mppc.alphacron.de/ fork from PPP project by Jan Dubiec # ppp-2.4.2 or later with MPPE and MPPC, kernel module ppp_mppe_mppc.o # {{{ # Require MPPE 128-bit encryption #mppe required,stateless # }}} IPtables is stopped on clients and server, Also net.ipv4.ip_forward = 1 is enabled on PPTP Server. How can i solve this problem .?

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• Sharing Bandwidth and Prioritizing Realtime Traffic via HTB, Which Scenario Works Better?

  - by Mecki

  I would like to add some kind of traffic management to our Internet line. After reading a lot of documentation, I think HFSC is too complicated for me (I don't understand all the curves stuff, I'm afraid I will never get it right), CBQ is not recommend, and basically HTB is the way to go for most people. Our internal network has three "segments" and I'd like to share bandwidth more or less equally between those (at least in the beginning). Further I must prioritize traffic according to at least three kinds of traffic (realtime traffic, standard traffic, and bulk traffic). The bandwidth sharing is not as important as the fact that realtime traffic should always be treated as premium traffic whenever possible, but of course no other traffic class may starve either. The question is, what makes more sense and also guarantees better realtime throughput: Creating one class per segment, each having the same rate (priority doesn't matter for classes that are no leaves according to HTB developer) and each of these classes has three sub-classes (leaves) for the 3 priority levels (with different priorities and different rates). Having one class per priority level on top, each having a different rate (again priority won't matter) and each having 3 sub-classes, one per segment, whereas all 3 in the realtime class have highest prio, lowest prio in the bulk class, and so on. I'll try to make this more clear with the following ASCII art image: Case 1: root --+--> Segment A | +--> High Prio | +--> Normal Prio | +--> Low Prio | +--> Segment B | +--> High Prio | +--> Normal Prio | +--> Low Prio | +--> Segment C +--> High Prio +--> Normal Prio +--> Low Prio Case 2: root --+--> High Prio | +--> Segment A | +--> Segment B | +--> Segment C | +--> Normal Prio | +--> Segment A | +--> Segment B | +--> Segment C | +--> Low Prio +--> Segment A +--> Segment B +--> Segment C Case 1 Seems like the way most people would do it, but unless I don't read the HTB implementation details correctly, Case 2 may offer better prioritizing. The HTB manual says, that if a class has hit its rate, it may borrow from its parent and when borrowing, classes with higher priority always get bandwidth offered first. However, it also says that classes having bandwidth available on a lower tree-level are always preferred to those on a higher tree level, regardless of priority. Let's assume the following situation: Segment C is not sending any traffic. Segment A is only sending realtime traffic, as fast as it can (enough to saturate the link alone) and Segment B is only sending bulk traffic, as fast as it can (again, enough to saturate the full link alone). What will happen? Case 1: Segment A-High Prio and Segment B-Low Prio both have packets to send, since A-High Prio has the higher priority, it will always be scheduled first, till it hits its rate. Now it tries to borrow from Segment A, but since Segment A is on a higher level and Segment B-Low Prio has not yet hit its rate, this class is now served first, till it also hits the rate and wants to borrow from Segment B. Once both have hit their rates, both are on the same level again and now Segment A-High Prio is going to win again, until it hits the rate of Segment A. Now it tries to borrow from root (which has plenty of traffic spare, as Segment C is not using any of its guaranteed traffic), but again, it has to wait for Segment B-Low Prio to also reach the root level. Once that happens, priority is taken into account again and this time Segment A-High Prio will get all the bandwidth left over from Segment C. Case 2: High Prio-Segment A and Low Prio-Segment B both have packets to send, again High Prio-Segment A is going to win as it has the higher priority. Once it hits its rate, it tries to borrow from High Prio, which has bandwidth spare, but being on a higher level, it has to wait for Low Prio-Segment B again to also hit its rate. Once both have hit their rate and both have to borrow, High Prio-Segment A will win again until it hits the rate of the High Prio class. Once that happens, it tries to borrow from root, which has again plenty of bandwidth left (all bandwidth of Normal Prio is unused at the moment), but it has to wait again until Low Prio-Segment B hits the rate limit of the Low Prio class and also tries to borrow from root. Finally both classes try to borrow from root, priority is taken into account, and High Prio-Segment A gets all bandwidth root has left over. Both cases seem sub-optimal, as either way realtime traffic sometimes has to wait for bulk traffic, even though there is plenty of bandwidth left it could borrow. However, in case 2 it seems like the realtime traffic has to wait less than in case 1, since it only has to wait till the bulk traffic rate is hit, which is most likely less than the rate of a whole segment (and in case 1 that is the rate it has to wait for). Or am I totally wrong here? I thought about even simpler setups, using a priority qdisc. But priority queues have the big problem that they cause starvation if they are not somehow limited. Starvation is not acceptable. Of course one can put a TBF (Token Bucket Filter) into each priority class to limit the rate and thus avoid starvation, but when doing so, a single priority class cannot saturate the link on its own any longer, even if all other priority classes are empty, the TBF will prevent that from happening. And this is also sub-optimal, since why wouldn't a class get 100% of the line's bandwidth if no other class needs any of it at the moment? Any comments or ideas regarding this setup? It seems so hard to do using standard tc qdiscs. As a programmer it was such an easy task if I could simply write my own scheduler (which I'm not allowed to do).

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• How to stop a ICMP attack?

  - by cumhur onat

  We are under a heavy icmp flood attack. Tcpdump shows the result below. Altough we have blocked ICMP with iptables tcpdump still prints icmp packets. I've also attached iptables configuration and "top" result. Is there any thing I can do to completely stop icmp packets? [root@server downloads]# tcpdump icmp -v -n -nn tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 03:02:47.810957 IP (tos 0x0, ttl 49, id 16007, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 124, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.811559 IP (tos 0x0, ttl 49, id 16010, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 52, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.811922 IP (tos 0x0, ttl 49, id 16012, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 122, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.812485 IP (tos 0x0, ttl 49, id 16015, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 126, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.812613 IP (tos 0x0, ttl 49, id 16016, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 122, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.812992 IP (tos 0x0, ttl 49, id 16018, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 122, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.813582 IP (tos 0x0, ttl 49, id 16020, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 52, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.814092 IP (tos 0x0, ttl 49, id 16023, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 120, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.814233 IP (tos 0x0, ttl 49, id 16024, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 120, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.815579 IP (tos 0x0, ttl 49, id 16025, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 50, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.815726 IP (tos 0x0, ttl 49, id 16026, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 IP (tos 0x0, ttl 50, id 31864, offset 0, flags [none], proto: ICMP (1), length: 76) 77.92.136.196 > 94.201.175.188: [|icmp] 03:02:47.815890 IP (tos 0x0, ttl 49, id 16027, offset 0, flags [none], proto: ICMP (1), length: 56) 80.227.64.183 > 77.92.136.196: ICMP redirect 94.201.175.188 to host 80.227.64.129, length 36 iptables configuration: [root@server etc]# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination ofis tcp -- anywhere anywhere tcp dpt:mysql ofis tcp -- anywhere anywhere tcp dpt:ftp DROP icmp -- anywhere anywhere Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination DROP icmp -- anywhere anywhere Chain ofis (2 references) target prot opt source destination ACCEPT all -- OUR_OFFICE_IP anywhere DROP all -- anywhere anywhere top: top - 03:12:19 up 400 days, 15:43, 3 users, load average: 1.49, 1.67, 2.61 Tasks: 751 total, 3 running, 748 sleeping, 0 stopped, 0 zombie Cpu(s): 8.2%us, 1.0%sy, 0.0%ni, 87.9%id, 2.1%wa, 0.1%hi, 0.7%si, 0.0%st Mem: 32949948k total, 26906844k used, 6043104k free, 4707676k buffers Swap: 10223608k total, 0k used, 10223608k free, 14255584k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 36 root 39 19 0 0 0 R 100.8 0.0 17:03.56 ksoftirqd/11 10552 root 15 0 11408 1460 676 R 5.7 0.0 0:00.04 top 7475 lighttpd 15 0 304m 22m 15m S 3.8 0.1 0:05.37 php-cgi 1294 root 10 -5 0 0 0 S 1.9 0.0 380:54.73 kjournald 3574 root 15 0 631m 11m 5464 S 1.9 0.0 0:00.65 node 7766 lighttpd 16 0 302m 19m 14m S 1.9 0.1 0:05.70 php-cgi 10237 postfix 15 0 52572 2216 1692 S 1.9 0.0 0:00.02 scache 1 root 15 0 10372 680 572 S 0.0 0.0 0:07.99 init 2 root RT -5 0 0 0 S 0.0 0.0 0:16.72 migration/0 3 root 34 19 0 0 0 S 0.0 0.0 0:00.06 ksoftirqd/0 4 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 0 0 0 S 0.0 0.0 1:10.46 migration/1 6 root 34 19 0 0 0 S 0.0 0.0 0:01.11 ksoftirqd/1 7 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 0 0 0 S 0.0 0.0 2:36.15 migration/2 9 root 34 19 0 0 0 S 0.0 0.0 0:00.19 ksoftirqd/2 10 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 0 0 0 S 0.0 0.0 3:48.91 migration/3 12 root 34 19 0 0 0 S 0.0 0.0 0:00.20 ksoftirqd/3 13 root RT -5 0 0 0 S 0.0 0.0 0:00.00 watchdog/3 uname -a [root@server etc]# uname -a Linux thisis.oursite.com 2.6.18-238.19.1.el5 #1 SMP Fri Jul 15 07:31:24 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux arp -an [root@server downloads]# arp -an ? (77.92.136.194) at 00:25:90:04:F0:90 [ether] on eth0 ? (192.168.0.2) at 00:25:90:04:F0:91 [ether] on eth1 ? (77.92.136.193) at 00:23:9C:0B:CD:01 [ether] on eth0

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• EC2 instance suddenly refusing SSH connections and won't respond to ping

  - by Chris

  My instance was running fine and this morning I was able to access a Ruby on Rails app hosted on it. An hour later I suddenly wasn't able to access my site, my SSH connection attempts were refused and the server wasn't even responding to ping. I didn't change anything on my system during that hour and reboots aren't fixing it. I've never had any problems connecting or pinging the system before. Can someone please help? This is on my production system! OS: CentOS 5 AMI ID: ami-10b55379 Type: m1.small [] ~% ssh -v *****@meeteor.com OpenSSH_5.2p1, OpenSSL 0.9.8l 5 Nov 2009 debug1: Reading configuration data /etc/ssh_config debug1: Connecting to meeteor.com [184.73.235.191] port 22. debug1: connect to address 184.73.235.191 port 22: Connection refused ssh: connect to host meeteor.com port 22: Connection refused [] ~% ping meeteor.com PING meeteor.com (184.73.235.191): 56 data bytes Request timeout for icmp_seq 0 Request timeout for icmp_seq 1 Request timeout for icmp_seq 2 ^C --- meeteor.com ping statistics --- 4 packets transmitted, 0 packets received, 100.0% packet loss [] ~% ========= System Log ========= Restarting system. Linux version 2.6.16-xenU ([email protected]) (gcc version 4.0.1 20050727 (Red Hat 4.0.1-5)) #1 SMP Mon May 28 03:41:49 SAST 2007 BIOS-provided physical RAM map: Xen: 0000000000000000 - 000000006a400000 (usable) 980MB HIGHMEM available. 727MB LOWMEM available. NX (Execute Disable) protection: active IRQ lockup detection disabled Built 1 zonelists Kernel command line: root=/dev/sda1 ro 4 Enabling fast FPU save and restore... done. Enabling unmasked SIMD FPU exception support... done. Initializing CPU#0 PID hash table entries: 4096 (order: 12, 65536 bytes) Xen reported: 2599.998 MHz processor. Dentry cache hash table entries: 131072 (order: 7, 524288 bytes) Inode-cache hash table entries: 65536 (order: 6, 262144 bytes) Software IO TLB disabled vmalloc area: ee000000-f53fe000, maxmem 2d7fe000 Memory: 1718700k/1748992k available (1958k kernel code, 20948k reserved, 620k data, 144k init, 1003528k highmem) Checking if this processor honours the WP bit even in supervisor mode... Ok. Calibrating delay using timer specific routine.. 5202.30 BogoMIPS (lpj=26011526) Mount-cache hash table entries: 512 CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 1024K (64 bytes/line) Checking 'hlt' instruction... OK. Brought up 1 CPUs migration_cost=0 Grant table initialized NET: Registered protocol family 16 Brought up 1 CPUs xen_mem: Initialising balloon driver. highmem bounce pool size: 64 pages VFS: Disk quotas dquot_6.5.1 Dquot-cache hash table entries: 1024 (order 0, 4096 bytes) Initializing Cryptographic API io scheduler noop registered io scheduler anticipatory registered (default) io scheduler deadline registered io scheduler cfq registered i8042.c: No controller found. RAMDISK driver initialized: 16 RAM disks of 4096K size 1024 blocksize Xen virtual console successfully installed as tty1 Event-channel device installed. netfront: Initialising virtual ethernet driver. mice: PS/2 mouse device common for all mice md: md driver 0.90.3 MAX_MD_DEVS=256, MD_SB_DISKS=27 md: bitmap version 4.39 NET: Registered protocol family 2 Registering block device major 8 IP route cache hash table entries: 65536 (order: 6, 262144 bytes) TCP established hash table entries: 262144 (order: 9, 2097152 bytes) TCP bind hash table entries: 65536 (order: 7, 524288 bytes) TCP: Hash tables configured (established 262144 bind 65536) TCP reno registered TCP bic registered NET: Registered protocol family 1 NET: Registered protocol family 17 NET: Registered protocol family 15 Using IPI No-Shortcut mode md: Autodetecting RAID arrays. md: autorun ... md: ... autorun DONE. kjournald starting. Commit interval 5 seconds EXT3-fs: mounted filesystem with ordered data mode. VFS: Mounted root (ext3 filesystem) readonly. Freeing unused kernel memory: 144k freed *************************************************************** *************************************************************** ** WARNING: Currently emulating unsupported memory accesses ** ** in /lib/tls glibc libraries. The emulation is ** ** slow. To ensure full performance you should ** ** install a 'xen-friendly' (nosegneg) version of ** ** the library, or disable tls support by executing ** ** the following as root: ** ** mv /lib/tls /lib/tls.disabled ** ** Offending process: init (pid=1) ** *************************************************************** *************************************************************** Pausing... 5Pausing... 4Pausing... 3Pausing... 2Pausing... 1Continuing... INIT: version 2.86 booting Welcome to CentOS release 5.4 (Final) Press 'I' to enter interactive startup. Setting clock : Fri Oct 1 14:35:26 EDT 2010 [ OK ] Starting udev: [ OK ] Setting hostname localhost.localdomain: [ OK ] No devices found Setting up Logical Volume Management: [ OK ] Checking filesystems Checking all file systems. [/sbin/fsck.ext3 (1) -- /] fsck.ext3 -a /dev/sda1 /dev/sda1: clean, 275424/1310720 files, 1161123/2621440 blocks [ OK ] Remounting root filesystem in read-write mode: [ OK ] Mounting local filesystems: [ OK ] Enabling local filesystem quotas: [ OK ] Enabling /etc/fstab swaps: [ OK ] INIT: Entering runlevel: 4 Entering non-interactive startup Starting background readahead: [ OK ] Applying ip6tables firewall rules: modprobe: FATAL: Module ip6_tables not found. ip6tables-restore v1.3.5: ip6tables-restore: unable to initializetable 'filter' Error occurred at line: 3 Try `ip6tables-restore -h' or 'ip6tables-restore --help' for more information. [FAILED] Applying iptables firewall rules: [ OK ] Loading additional iptables modules: ip_conntrack_netbios_ns [ OK ] Bringing up loopback interface: [ OK ] Bringing up interface eth0: Determining IP information for eth0... done. [ OK ] Starting auditd: [FAILED] Starting irqbalance: [ OK ] Starting portmap: [ OK ] FATAL: Module lockd not found. Starting NFS statd: [ OK ] Starting RPC idmapd: FATAL: Module sunrpc not found. FATAL: Error running install command for sunrpc Error: RPC MTAB does not exist. Starting system message bus: [ OK ] Starting Bluetooth services:[ OK ] [ OK ] Can't open RFCOMM control socket: Address family not supported by protocol Mounting other filesystems: [ OK ] Starting PC/SC smart card daemon (pcscd): [ OK ] Starting hidd: Can't open HIDP control socket: Address family not supported by protocol [FAILED] Starting autofs: Starting automount: automount: test mount forbidden or incorrect kernel protocol version, kernel protocol version 5.00 or above required. [FAILED] [FAILED] Starting sshd: [ OK ] Starting cups: [ OK ] Starting sendmail: [ OK ] Starting sm-client: [ OK ] Starting console mouse services: no console device found[FAILED] Starting crond: [ OK ] Starting xfs: [ OK ] Starting anacron: [ OK ] Starting atd: [ OK ] % Total % Received % Xferd Average Speed Time Time Time Current Dload Upload Total Spent Left Speed 100 390 100 390 0 0 58130 0 --:--:-- --:--:-- --:--:-- 58130 100 390 100 390 0 0 56984 0 --:--:-- --:--:-- --:--:-- 0 Starting yum-updatesd: [ OK ] Starting Avahi daemon... [ OK ] Starting HAL daemon: [ OK ] Starting OSSEC: [ OK ] Starting smartd: [ OK ] c CentOS release 5.4 (Final) Kernel 2.6.16-xenU on an i686 domU-12-31-39-00-C4-97 login: INIT: Id "2" respawning too fast: disabled for 5 minutes INIT: Id "3" respawning too fast: disabled for 5 minutes INIT: Id "4" respawning too fast: disabled for 5 minutes INIT: Id "5" respawning too fast: disabled for 5 minutes INIT: Id "6" respawning too fast: disabled for 5 minutes

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• Anyone had any experience with *.pcap manipulation libs?

  - by zxcvbnm

  I'm using the SharpPcap + PacketDotNet libraries to process some .pcap files and came across a bug in the way the timestamps are calculated. Take this Timeval property, which is something along these lines: PosixTimeval Timeval { DateTime Date; ulong Seconds; ulong MicroSeconds; } The problem is as follows: Suppose you have a trace open in Wireshark with one of the packets with a timestamp of "0.002". Once you open it within one of your programs, it retrieves the packet and its Timeval is setup such that Seconds = 0 and MicroSeconds = 002 = 2. This is done under the hood, so there is no way to avoid it as far as I can tell. My question is if that problem is common to other libraries (and maybe all of them?) who manipulate the pcap file format, which I think are built around the same collection of c/c++ functions, or if this is a problem only with the ones I'm using.

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• Flow diagram in html/css

  - by viraptor

  Hi, is there some good way to create a flow / swimline diagram without resorting to scripting or tables? I'm talking about showing different hosts sending packets (so hosts on the X axis, time on the Y and arrows to the destination host). It seems like too much elements for tables (especially the arrows spanning multiple columns either way), but on the other hand, divs would be hard to position in the right place horizontally (they'd have to be basically aligned to a specified "column"). Is there any good way out? Any helper frameworks? (I don't want to do canvas stuff unless really needed)

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• Unexpected behavior with AudioQueueServices callback while recording audio

  - by rcw3

  I'm recording a continuous stream of data using AudioQueueServices. It is my understanding that the callback will only be called when the buffer fills with data. In practice, the first callback has a full buffer, the 2nd callback is 3/4 full, the 3rd callback is full, the 4th is 3/4 full, and so on. These buffers are 8000 packets (recording 8khz audio) - so I should be getting back 1s of audio to the callback each time. I've confirmed that my audio queue buffer size is correct (and is somewhat confirmed by the behavior). What am I doing wrong? Should I be doing something in the AudioQueueNewInput with a different RunLoop? I tried but this didn't seem to make a difference... By the way, if I run in the debugger, each callback is full with 8000 samples - making me think this is a threading / timing thing.

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• Reusable socket

  - by Andrew

  I tryed to create a socket in php and reuse it from other process. I know this can be done with a daemon script but I want to do this without. I created a socket and binded it to a specific port. $sock = socket_create (AF_INET, SOCK_STREAM, SOL_TCP); socket_set_option ($sock, SOL_SOCKET, SO_REUSEADDR, 1); socket_bind ($sock, 'xx.xx.xx.xx', 10000); socket_connect ($sock, $host, $port); And from another php file I did the same thing. But the packets that I send from the 2 file are not "validated" by host. I sniffed all ports and I see that it uses same local and destination port. I don't understand where is the problem. Can you help me with this? It's ok in any other programming language, or any other solution for this. Andrew

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• libpcap IP Packet Reassembly

  - by holydiver

  Hello all, I'm looking for a sample code for IP packet reassembly in C with libpcap*. Is IP packet defragmentation implemented in libpcap library officially? I've found this proposal : http://www.mail-archive.com/[email protected]/msg02991.html[this][1] . Are there any implementation of defragmentation of IP packets. [1]: http://www.mail-archive.com/[email protected]/msg02991.html [+] *The reason for looking for the sample code is just for fun and learning and i don't want to spend too much time for writing code for IP packet reassembly.

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• linux: disable using loopback and send data via wire between 2 eth cards of one comp

  - by osgx

  Hello I have a comp with 2 eth cards, connected with patch-cord (direct eth. cable from 1st to 2nd). The linux is installed, I want to send data from 1st network card to 2nd. And I want to force the packet to pass via cable. I can set up any ip on cards. With ping I get counters on cards constant. Is it possible with tcp/ip sockets? PS. I need to write a program. which will send packets via eth, so stackoverflow-related question. There can be some OS-dependent way, they will help me too

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• What happens after a packet is captured?

  - by Rayne

  Hi all, I've been reading about what happens after packets are captured by NICs, and the more I read, the more I'm confused. Firstly, I've read that traditionally, after a packet is captured by the NIC, it gets copied to a block of memory in the kernel space, then to the user space for whatever application that then works on the packet data. Then I read about DMA, where the NIC directly copies the packet into memory, bypassing the CPU. So is the NIC - kernel memory - User space memory flow still valid? Also, do most NIC (e.g. Myricom) use DMA to improve packet capture rates? Secondly, does RSS (Receive Side Scaling) work similarly in both Windows and Linux systems? I can only find detailed explanations on how RSS works in MSDN articles, where they talk about how RSS (and MSI-X) works on Windows Server 2008. But the same concept of RSS and MSI-X should still apply for linux systems, right? Thank you. Regards, Rayne

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• synchronizing audio over a network

  - by sharkin

  I'm in startup of designing a client/server audio system which can stream audio arbitrarily over a network. One central server pumps out an audio stream and x number of clients receives the audio data and plays it. So far no magic needed and I have even got this scenario to work with VLC media player out of the box. However, the tricky part seems to be synchronizing the audio playback so that all clients are in audible synch (actual latency can be allowed as long as it is perceived to be in sync by a human listener). My question is if there's any known method or algorithm to use for these types of synchronization problems (video is probably solved the same way). My own initial thoughts centers around synchronizing clocks between physical machines and thereby creating a virtual "main timer" and somehow aligning audio data packets against it. Some products already solving the problem: http://www.sonos.com http://netchorus.com/ Any pointers are most welcome. Thanks. PS: This related question seem to have died long ago.

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• Video packet capture over multiple IP cameras

  - by nimals1986

  Hello We are working on a C language application which is simple RTSP/RTP client to record video from Axis a number of Cameras . We launch a pthread for each of the camera which establishes the RTP session and begins to record the packets captured suing the recvfrom() call... A single camera single pthread records fine for well over a day without issues.. but testing with more cameras available,about 25(so 25 pthreads), the recording to file goes fine for like 15 to 20 mins and then the recording just stops ..the application still keeps running .. Its been over a month and a half we have been trying with varied implementations but nothing seems to help .. Please provide suggestions.. We are using CentOS 5 platform

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• Conceal packet loss in PCM stream

  - by ZeroDefect

  I am looking to use 'Packet Loss Concealment' to conceal lost PCM frames in an audio stream. Unfortunately, I cannot find a library that is accessible without all the licensing restrictions and code bloat (...up for some suggestions though). I have located some GPL code written by Steve Underwood for the Asterisk project which implements PLC. There are several limitations; although, as Steve suggests in his code, his algorithm can be applied to different streams with a bit of work. Currently, the code works with 8kHz 16-bit signed mono streams. Variations of the code can be found through a simple search of Google Code Search. My hope is that I can adapt the code to work with other streams. Initially, the goal is to adjust the algorithm for 8+ kHz, 16-bit signed, multichannel audio (all in a C++ environment). Eventually, I'm looking to make the code available under the GPL license in hopes that it could be of benefit to others... Attached is the code below with my efforts. The code includes a main function that will "drop" a number of frames with a given probability. Unfortunately, the code does not quite work as expected. I'm receiving EXC_BAD_ACCESS when running in gdb, but I don't get a trace from gdb when using 'bt' command. Clearly, I'm trampimg on memory some where but not sure exactly where. When I comment out the *amdf_pitch* function, the code runs without crashing... int main (int argc, char *argv[]) { std::ifstream fin("C:\\cc32kHz.pcm"); if(!fin.is_open()) { std::cout << "Failed to open input file" << std::endl; return 1; } std::ofstream fout_repaired("C:\\cc32kHz_repaired.pcm"); if(!fout_repaired.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } std::ofstream fout_lossy("C:\\cc32kHz_lossy.pcm"); if(!fout_lossy.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } audio::PcmConcealer Concealer; Concealer.Init(1, 16, 32000); //Generate random numbers; srand( time(NULL) ); int value = 0; int probability = 5; while(!fin.eof()) { char arr[2]; fin.read(arr, 2); //Generate's random number; value = rand() % 100 + 1; if(value <= probability) { char blank[2] = {0x00, 0x00}; fout_lossy.write(blank, 2); //Fill in data; Concealer.Fill((int16_t *)blank, 1); fout_repaired.write(blank, 2); } else { //Write data to file; fout_repaired.write(arr, 2); fout_lossy.write(arr, 2); Concealer.Receive((int16_t *)arr, 1); } } fin.close(); fout_repaired.close(); fout_lossy.close(); return 0; } PcmConcealer.hpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #ifndef __PCMCONCEALER_HPP__ #define __PCMCONCEALER_HPP__ /** 1. What does it do? The packet loss concealment module provides a suitable synthetic fill-in signal, to minimise the audible effect of lost packets in VoIP applications. It is not tied to any particular codec, and could be used with almost any codec which does not specify its own procedure for packet loss concealment. Where a codec specific concealment procedure exists, the algorithm is usually built around knowledge of the characteristics of the particular codec. It will, therefore, generally give better results for that particular codec than this generic concealer will. 2. How does it work? While good packets are being received, the plc_rx() routine keeps a record of the trailing section of the known speech signal. If a packet is missed, plc_fillin() is called to produce a synthetic replacement for the real speech signal. The average mean difference function (AMDF) is applied to the last known good signal, to determine its effective pitch. Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech will be repeated over and over until the real speech resumes. However, several refinements are needed to obtain smooth pleasant sounding results. - The two ends of the stored cycle of speech will not always fit together smoothly. This can cause roughness, or even clicks, at the joins between cycles. To soften this, the 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e. in total, the last 5/4 pitch periods of real speech are used). - The start of the synthetic speech will not always fit together smoothly with the tail of real speech passed on before the erasure was identified. Ideally, we would like to modify the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However, it is too late for that. We could have delayed the real speech a little, but that would require more buffer manipulation, and hurt the efficiency of the no-lost-packets case (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed, and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result seems quite acceptable. - As we progress into the erasure, the chances of the synthetic signal being anything like correct steadily fall. Therefore, the volume of the synthesized signal is made to decay linearly, such that after 50ms of missing audio it is reduced to silence. - When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the start of the real speech. If the erasure is small, this smoothes the transition. If the erasure is long, and the synthetic signal has faded to zero, the blending softens the start up of the real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. 3. How do I use it? Before audio is processed, call plc_init() to create an instance of the packet loss concealer. For each received audio packet that is acceptable (i.e. not including those being dropped for being too late) call plc_rx() to record the content of the packet. Note this may modify the packet a little after a period of packet loss, to blend real synthetic data smoothly. When a real packet is not available in time, call plc_fillin() to create a sythetic substitute. That's it! */ /*! Minimum allowed pitch (66 Hz) */ #define PLC_PITCH_MIN(SAMPLE_RATE) ((double)(SAMPLE_RATE) / 66.6) /*! Maximum allowed pitch (200 Hz) */ #define PLC_PITCH_MAX(SAMPLE_RATE) ((SAMPLE_RATE) / 200) /*! Maximum pitch OLA window */ //#define PLC_PITCH_OVERLAP_MAX(SAMPLE_RATE) ((PLC_PITCH_MIN(SAMPLE_RATE)) >> 2) /*! The length over which the AMDF function looks for similarity (20 ms) */ #define CORRELATION_SPAN(SAMPLE_RATE) ((20 * (SAMPLE_RATE)) / 1000) /*! History buffer length. The buffer must also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment. */ //#define PLC_HISTORY_LEN(SAMPLE_RATE) ((CORRELATION_SPAN(SAMPLE_RATE)) + (PLC_PITCH_MIN(SAMPLE_RATE))) namespace audio { typedef struct { /*! Consecutive erased samples */ int missing_samples; /*! Current offset into pitch period */ int pitch_offset; /*! Pitch estimate */ int pitch; /*! Buffer for a cycle of speech */ float *pitchbuf;//[PLC_PITCH_MIN]; /*! History buffer */ short *history;//[PLC_HISTORY_LEN]; /*! Current pointer into the history buffer */ int buf_ptr; } plc_state_t; class PcmConcealer { public: PcmConcealer(); ~PcmConcealer(); void Init(int channels, int bit_depth, int sample_rate); //Process a block of received audio samples. int Receive(short amp[], int frames); //Fill-in a block of missing audio samples. int Fill(short amp[], int frames); void Destroy(); private: int amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames); void save_history(plc_state_t *s, short *buf, int channel_index, int frames); void normalise_history(plc_state_t *s); /** Holds the states of each of the channels **/ std::vector< plc_state_t * > ChannelStates; int plc_pitch_min; int plc_pitch_max; int plc_pitch_overlap_max; int correlation_span; int plc_history_len; int channel_count; int sample_rate; bool Initialized; }; } #endif PcmConcealer.cpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #include "audio/PcmConcealer.hpp" /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */ #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ #if !defined(INT16_MAX) #define INT16_MAX (32767) #define INT16_MIN (-32767-1) #endif #ifdef WIN32 inline double rint(double x) { return floor(x + 0.5); } #endif inline short fsaturate(double damp) { if (damp > 32767.0) return INT16_MAX; if (damp < -32768.0) return INT16_MIN; return (short)rint(damp); } namespace audio { PcmConcealer::PcmConcealer() : Initialized(false) { } PcmConcealer::~PcmConcealer() { Destroy(); } void PcmConcealer::Init(int channels, int bit_depth, int sample_rate) { if(Initialized) return; if(channels <= 0 || bit_depth != 16) return; Initialized = true; channel_count = channels; this->sample_rate = sample_rate; ////////////// double min = PLC_PITCH_MIN(sample_rate); int imin = (int)min; double max = PLC_PITCH_MAX(sample_rate); int imax = (int)max; plc_pitch_min = imin; plc_pitch_max = imax; plc_pitch_overlap_max = (plc_pitch_min >> 2); correlation_span = CORRELATION_SPAN(sample_rate); plc_history_len = correlation_span + plc_pitch_min; ////////////// for(int i = 0; i < channel_count; i ++) { plc_state_t *t = new plc_state_t; memset(t, 0, sizeof(plc_state_t)); t->pitchbuf = new float[plc_pitch_min]; t->history = new short[plc_history_len]; ChannelStates.push_back(t); } } void PcmConcealer::Destroy() { if(!Initialized) return; while(ChannelStates.size()) { plc_state_t *s = ChannelStates.at(0); if(s) { if(s->history) delete s->history; if(s->pitchbuf) delete s->pitchbuf; memset(s, 0, sizeof(plc_state_t)); delete s; } ChannelStates.erase(ChannelStates.begin()); } ChannelStates.clear(); Initialized = false; } //Process a block of received audio samples. int PcmConcealer::Receive(short amp[], int frames) { if(!Initialized) return 0; int j = 0; for(int k = 0; k < ChannelStates.size(); k++) { int i; int overlap_len; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples) { /* Although we have a real signal, we need to smooth it to fit well with the synthetic signal we used for the previous block */ /* The start of the real data is overlapped with the next 1/4 cycle of the synthetic data. */ pitch_overlap = s->pitch >> 2; if (pitch_overlap > frames) pitch_overlap = frames; gain = 1.0 - s->missing_samples * ATTENUATION_INCREMENT; if (gain < 0.0) gain = 0.0; new_step = 1.0/pitch_overlap; old_step = new_step*gain; new_weight = new_step; old_weight = (1.0 - new_step)*gain; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[index]); if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->missing_samples = 0; } save_history(s, amp, j, frames); j++; } return frames; } //Fill-in a block of missing audio samples. int PcmConcealer::Fill(short amp[], int frames) { if(!Initialized) return 0; int j =0; for(int k = 0; k < ChannelStates.size(); k++) { short *tmp = new short[plc_pitch_overlap_max]; int i; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; short *orig_amp; int orig_len; orig_amp = amp; orig_len = frames; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples == 0) { // As the gap in real speech starts we need to assess the last known pitch, //and prepare the synthetic data we will use for fill-in normalise_history(s); s->pitch = amdf_pitch(plc_pitch_min, plc_pitch_max, s->history + plc_history_len - correlation_span - plc_pitch_min, j, correlation_span); // We overlap a 1/4 wavelength pitch_overlap = s->pitch >> 2; // Cook up a single cycle of pitch, using a single of the real signal with 1/4 //cycle OLA'ed to make the ends join up nicely // The first 3/4 of the cycle is a simple copy for (i = 0; i < s->pitch - pitch_overlap; i++) s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]; // The last 1/4 of the cycle is overlapped with the end of the previous cycle new_step = 1.0/pitch_overlap; new_weight = new_step; for ( ; i < s->pitch; i++) { s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]*(1.0 - new_weight) + s->history[plc_history_len - 2*s->pitch + i]*new_weight; new_weight += new_step; } // We should now be ready to fill in the gap with repeated, decaying cycles // of what is in pitchbuf // We need to OLA the first 1/4 wavelength of the synthetic data, to smooth // it into the previous real data. To avoid the need to introduce a delay // in the stream, reverse the last 1/4 wavelength, and OLA with that. gain = 1.0; new_step = 1.0/pitch_overlap; old_step = new_step; new_weight = new_step; old_weight = 1.0 - new_step; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->history[plc_history_len - 1 - i] + new_weight * s->pitchbuf[i]); new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->pitch_offset = i; } else { gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; i = 0; } for ( ; gain > 0.0 && i < frames; i++) { int index = (i * channel_count) + j; amp[index] = s->pitchbuf[s->pitch_offset]*gain; gain -= ATTENUATION_INCREMENT; if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; } for ( ; i < frames; i++) { int index = (i * channel_count) + j; amp[i] = 0; } s->missing_samples += orig_len; save_history(s, amp, j, frames); delete [] tmp; j++; } return frames; } void PcmConcealer::save_history(plc_state_t *s, short *buf, int channel_index, int frames) { if (frames >= plc_history_len) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ //memcpy(s->history, buf + len - plc_history_len, sizeof(short)*plc_history_len); int frames_to_copy = plc_history_len; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + frames - plc_history_len)) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = 0; return; } if (s->buf_ptr + frames > plc_history_len) { /* Wraps around - must break into two sections */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*(plc_history_len - s->buf_ptr)); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = plc_history_len - s->buf_ptr; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } frames -= (plc_history_len - s->buf_ptr); //memcpy(s->history, buf + (plc_history_len - s->buf_ptr), sizeof(short)*len); frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + (plc_history_len - s->buf_ptr))) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = frames; return; } /* Can use just one section */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*len); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } s->buf_ptr += frames; } void PcmConcealer::normalise_history(plc_state_t *s) { short *tmp = new short[plc_history_len]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(short)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(short)*(plc_history_len - s->buf_ptr)); memcpy(s->history + plc_history_len - s->buf_ptr, tmp, sizeof(short)*s->buf_ptr); s->buf_ptr = 0; delete [] tmp; } int PcmConcealer::amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames) { int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < frames; j++) { int index1 = (channel_count * (i+j)) + channel_index; int index2 = (channel_count * j) + channel_index; //std::cout << "Index 1: " << index1 << ", Index 2: " << index2 << std::endl; acc += abs(amp[index1] - amp[index2]); } if (acc < min_acc) { min_acc = acc; pitch = i; } } std::cout << "Pitch: " << pitch << std::endl; return pitch; } } P.S. - I must confess that digital audio is not my forte...

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• How to recover gracefully from a C# udp socket exception

  - by Gearoid Murphy

  Context: I'm porting a linux perl app to C#, the server listens on a udp port and maintains multiple concurrent dialogs with remote clients via a single udp socket. During testing, I send out high volumes of packets to the udp server, randomly restarting the clients to observe the server registering the new connections. The problem is this: when I kill a udp client, there may still be data on the server destined for that client. When the server tries to send this data, it gets an icmp "no service available" message back and consequently an exception occurs on the socket. I cannot reuse this socket, when I try to associate a C# async handler with the socket, it complains about the exception, so I have to close and reopen the udp socket on the server port. Is this the only way around this problem?, surely there's some way of "fixing" the udp socket, as technically, UDP sockets shouldn't be aware of the status of a remote socket? Any help or pointers would be much appreciated. Thanks.

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• MPI: is there mpi libraries capable of message compression?

  - by osgx

  Sometimes MPI is used to send low-entropy data in messages. So it can be useful to try to compress messages before sending it. I know that MPI can work on very fast networks (10 Gbit/s and more), but many MPI programs are used with cheap network like 0,1G or 1Gbit/s Ethernet and with cheap (slow, low bisection) network switch. There is a very fast Snappy (wikipedia) compression algorithm, which has Compression speed is 250 MB/s and decompression speed is 500 MB/s so on compressible data and slow network it will give some speedup. Is there any MPI library which can compress MPI messages (at layer of MPI; not the compression of ip packets like in PPP). MPI messages are also structured, so there can be some special method, like compression of exponent part in array of double.

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• From admin to dev [closed]

  - by Terry

  Recently a friend of mine had gone from a high level NOC position to a developer. Before that he was just doing the help desk stuff. He has no degree, only the usual MIS/networking certifications and as far as I know only tinkers with code on the weekends. I can see where in some scenarios having a good understanding of configurations, packets, users, OU's, etc would be extremely beneficial to a developer. My question is this, how many full time developers started off this way? Even how many people dual wield the responsibility of developer/systems administrator/network administration?

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• Can't authenticate with different NTLM credentials in one session with java.net.URLConnection

  - by ndn

  When I access a HTTP server using the standard Java API (java.net.URLConnection), the credentials are "cached" after the first successful authentication, and subsequent calls to Authenticator.setDefault() have no effect. So, I need to restart the application in order to use different credentials. I don't observe this effect when Basic Authentication is used. But I need to use NTLM for the server I'm accessing, and the Jakarta Commons HttpClient isn't an alternative either because it doesn't support NTLMv2 (see http://oaklandsoftware.com/papers/ntlm.html) Looking at the packets using Wireshark, I also observe that before the first successful authentication, an authentication with the current Windows credentials is attempted first. But after succesful authentication, only the saved credentials are used. Is there any way to reset or change the credentials java.net.Authenticator is using after a successful NTLM authentication?

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