Search Results

Search found 1822 results on 73 pages for 'bandwidth caps'.

Page 20/73 | < Previous Page | 16 17 18 19 20 21 22 23 24 25 26 27  | Next Page >

  • Why does use of H264 in sender/receiver pipelines introduce just HUGE delay?

    - by Serguey Zefirov
    When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong. I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex: This is sender pipeline: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 Receiver pipeline: !/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time. Then I changed pipelines to use H264 along the video path. The sender becomes: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 And receiver becomes: #!/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

    Read the article

  • Customizing UISlider look

    - by tcurdt
    To customize the visual look of a UISlider you can set the thumb and track images. Part of the track images gets stretched to the appropriate with. From the documentation: A stretchable region sits between two end cap regions. The end caps define the portions of the image that remain as is and are not stretched. The stretchable region is a 1-point wide area between the end caps that can be replicated to make the image appear longer. Now the problem I have is that my stretchable region needs to be more than 1-point wide. (It's a pattern) Unfortunately the 1-point width seems to be hard coded in the SDK. Anyone having an idea how to work around this? Or will I have to write my own slider from scratch for this?

    Read the article

  • Movable Type: MTEntries sort_by="title" doesn't really work

    - by kohei
    Hello Im trying to sort <MTEntries> by title. I know you can use <MTEntries sort_by="title" sort_order="ascend"> but this modifier some how prioritizes capitalized letters first to the sort. Im not sure if this is a glitch in the system but this modifier should sort by purely the alphabets(caps or no caps) used in the title. Example: I would like to sort these titles alphabetically: APRICOT Aligator ABBEY Apple If <MTEntries sort_by="title" sort_order="ascend"> is used: ABBEY APRICOT Aligator Apple But it really should be (and I want) ABBEY Aligator Apple APRICOT Would someone know how to achive this?

    Read the article

  • FFmpeg Video Hosting for Linux and Windows Server

    - by Aditi
    FFmpeg hosting is a special type of web hosting where the host servers have video transcoding software loaded on them, which allows the automatic conversion of videos from one format to another. FFmpeg is a cross-platform solution for recording, converting, transcoding and stream audio and video. It includes libavcodec – the leading audio/video codec library. FFmpeg hosting gets its name from a set of server side programs (modules) called FFmpeg. There are a number of applications or web scripts available, which allow webmasters to create their own video sharing websites. Video hosting typically requires: PHP 4.3 and above (including support of CLI) Mencoder and also Mplayer FFMpeg-PHP MySQL database server LAME MP3 Encoder Libogg + Libvorbis GD Library 2 or higher CGI-BIN There are number of web service providers who provide FFmpeg hosting service. Following is a list of some of the Best FFmpeg hosting providers for both Linux and Windows Server below. Dream Host Dreamhost provides for web based email access, mail filtering, spam filtering, unlimited email ids, vacation autoresponder, python support, full CGI access and many more services. Price: $7.95 View Details Micfo It offers unlimited disk space and bandwidth. Other services include free domain for life and free Website Transfer with many more services. All in all one of the best option to consider. Price: $5 View Details Host Upon HostUpon offers FFMpeg Hosting on all their hosting packages, with readily installed modules to start a Video website or Social Network with Video uploading. These scripts such as Boonex Dolphin / PHPMotion / Social Engine / ABKsoft Scripts / Joomla Video Plugin / Clipshare / ClipBucket / Social Media / Rayzz / Vidi Script work with their ffmpeg. Their FFMPEG hosting plan offers 24/7/365 support with typical response time of 15min or less. Price: $5.95 View Details DownTown Host DownTown Host provides full and exceptional support by live chat and telephone. It has high-power, modern servers and the finest web server technology. It offers free search engine Submission and continuous data backup protection with free email forwarding and site move. There are many more services too. Site5 This ffmpeg service provider offers uptime guarantee, a real time stats on each server and many more attractive services. Price: $4.95 View Details Cirtex Hosting Cirtex Hosting allows to host 7 websites & domains and provides for unlimited storage space and monthly bandwidth. It also offers FTP and email accounts and many more services. Price: $2.49 View Details FLV Hosting FLV hosting supplies RTMP SERVER STREAMING for large size video streaming and server side recording. It is flexible and costs less. They customize to the clients requirements. Price: $9.95 View Details AptHost This hosting service provides for 24x7x365 Premium Support and fully ffmpeg enabled services. Price: $4.95 View Details HostMDS Great Support, Priced Low. It provides for SSH access, CGI, Ruby on Rails, Perl, PHP, MySQL, front page extentions, 24/7 Support, FREE Domain transfer and spam filtering. It offers instant account setup, low latency fast bandwidth & much more! They were formerly known as Vistapages. Price: $4.95 View Details Related posts:Best WordPress Video Themes for a Video Blog Free Web Based Applications 24+ Coda Alternatives for Windows and Linux

    Read the article

  • Exploring TCP throughput with DTrace

    - by user12820842
    One key measure to use when assessing TCP throughput is assessing the amount of unacknowledged data in the pipe. This is sometimes termed the Bandwidth Delay Product (BDP) (note that BDP is often used more generally as the product of the link capacity and the end-to-end delay). In DTrace terms, the amount of unacknowledged data in bytes for the connection is the different between the next sequence number to send and the lowest unacknoweldged sequence number (tcps_snxt - tcps_suna). According to the theory, when the number of unacknowledged bytes for the connection is less than the receive window of the peer, the path bandwidth is the limiting factor for throughput. In other words, if we can fill the pipe without the peer TCP complaining (by virtue of its window size reaching 0), we are purely bandwidth-limited. If the peer's receive window is too small however, the sending TCP has to wait for acknowledgements before it can send more data. In this case the round-trip time (RTT) limits throughput. In such cases the effective throughput limit is the window size divided by the RTT, e.g. if the window size is 64K and the RTT is 0.5sec, the throughput is 128K/s. So a neat way to visually determine if the receive window of clients may be too small should be to compare the distribution of BDP values for the server versus the client's advertised receive window. If the BDP distribution overlaps the send window distribution such that it is to the right (or lower down in DTrace since quantizations are displayed vertically), it indicates that the amount of unacknowledged data regularly exceeds the client's receive window, so that it is possible that the sender may have more data to send but is blocked by a zero-window on the client side. In the following example, we compare the distribution of BDP values to the receive window advertised by the receiver (10.175.96.92) for a large file download via http. # dtrace -s tcp_tput.d ^C BDP(bytes) 10.175.96.92 80 value ------------- Distribution ------------- count -1 | 0 0 | 6 1 | 0 2 | 0 4 | 0 8 | 0 16 | 0 32 | 0 64 | 0 128 | 0 256 | 3 512 | 0 1024 | 0 2048 | 9 4096 | 14 8192 | 27 16384 | 67 32768 |@@ 1464 65536 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ 32396 131072 | 0 SWND(bytes) 10.175.96.92 80 value ------------- Distribution ------------- count 16384 | 0 32768 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ 17067 65536 | 0 Here we have a puzzle. We can see that the receiver's advertised window is in the 32768-65535 range, while the amount of unacknowledged data in the pipe is largely in the 65536-131071 range. What's going on here? Surely in a case like this we should see zero-window events, since the amount of data in the pipe regularly exceeds the window size of the receiver. We can see that we don't see any zero-window events since the SWND distribution displays no 0 values - it stays within the 32768-65535 range. The explanation is straightforward enough. TCP Window scaling is in operation for this connection - the Window Scale TCP option is used on connection setup to allow a connection to advertise (and have advertised to it) a window greater than 65536 bytes. In this case the scaling shift is 1, so this explains why the SWND values are clustered in the 32768-65535 range rather than the 65536-131071 range - the SWND value needs to be multiplied by two since the reciever is also scaling its window by a shift factor of 1. Here's the simple script that compares BDP and SWND distributions, fixed to take account of window scaling. #!/usr/sbin/dtrace -s #pragma D option quiet tcp:::send / (args[4]-tcp_flags & (TH_SYN|TH_RST|TH_FIN)) == 0 / { @bdp["BDP(bytes)", args[2]-ip_daddr, args[4]-tcp_sport] = quantize(args[3]-tcps_snxt - args[3]-tcps_suna); } tcp:::receive / (args[4]-tcp_flags & (TH_SYN|TH_RST|TH_FIN)) == 0 / { @swnd["SWND(bytes)", args[2]-ip_saddr, args[4]-tcp_dport] = quantize((args[4]-tcp_window)*(1 tcps_snd_ws)); } And here's the fixed output. # dtrace -s tcp_tput_scaled.d ^C BDP(bytes) 10.175.96.92 80 value ------------- Distribution ------------- count -1 | 0 0 | 39 1 | 0 2 | 0 4 | 0 8 | 0 16 | 0 32 | 0 64 | 0 128 | 0 256 | 3 512 | 0 1024 | 0 2048 | 4 4096 | 9 8192 | 22 16384 | 37 32768 |@ 99 65536 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ 3858 131072 | 0 SWND(bytes) 10.175.96.92 80 value ------------- Distribution ------------- count 512 | 0 1024 | 1 2048 | 0 4096 | 2 8192 | 4 16384 | 7 32768 | 14 65536 |@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ 1956 131072 | 0

    Read the article

  • Can someone explain the true landscape of Rails vs PHP deployment, particularly within the context of Reseller-based web hosting (e.g., Hostgator)?

    - by rcd
    Currently, I have a reseller account with the company HostGator. I design websites, which up until now have occasionally been wrapped in Wordpress CMSs and the like (PHP applications). I then sell hosting (of the site I've designed) to the client, which is pretty simple, in that I can simply click a button and add a new shared hosting account/site with whatever settings I want. Furthermore, I then utilize WHMCS to automate billing and account management. It's a nice package and pretty simple. I pay something like $25 a month, and can sell a hundred accounts under this (because my clients bandwidth requirements are low). Now I am finding the need to develop more customized applications, including a minimalist CMS and several proprietary things. I soon anticipate developing these apps for clients as well. Thus, I've spent the past few months learning Rails, and it's coming along well now. The thing that has nagged at me all along, though, is the deployment issue. I can't wrap my brain around it. It seems like all of the popular options (Heroku, etc) have nice automation with git and are set up in the "Rails Way". I get that (sort of). But it's terribly expensive... a single dyno, a helper, and the cheapest database (which they say is mainly suitable for testing) that isn't limited to 5MB runs $51. This is for ONE app!!! Throw in a "production" DB and you're over $200. This is like... the same prices as getting a server somewhere, right? Meanwhile, going back to what I guess is a "traditional" hosting environment with Hostgator, their server only has Ruby 1.8.7 and Rails 2.3.5... No Rails 3. AND, no Passenger (not that I really understand the difference in CGI or mod_rails or whatever, but they say Passenger is the simplest). So I'm to understand that if I build an app in Rails 3, it won't run at all on this host? But damn, I already have these accounts under my reseller account there, all running static html and/or PHP stuff, right? So what now? How do I get all of this under one simple (and affordable) roof? Forgive my ignorance, but I just don't get it. Managing a VPS is cool and all, but entails learning server admin stuff and security... And it's expensive. I get that a shared and/or reseller "server-based" (forgive the terminology) may be inadequate for large-scale apps that use a lot of bandwidth... But what about for those of us who are building real (but small and low bandwidth) apps (with Rails) and who want to deploy them simply, cheaply, using the same conceptual approach as PHP? Even after learning all of this Ruby and Rails stuff for months, I'm questioning whether it's worth it when it comes to deployment. I want to build a small app, upload it to my home directory on a shared server account, and just make it run. Why should that be so hard? Am I just choosing the wrong language/framework? Forgive my ignorance in the subject; these questions are not rhetorical; just trying to learn here. So: 1) I'd appreciate if someone could give me a good rundown of how to understand deployment in Rails vs. PHP. 2) I'd appreciate if someone could address my issue with running a hosting/web business around reseller hosting (Hostgator) while also being able to host Rails apps. Can it be done? And how can a company like Hostgator completely ignore what's current in Rails/Ruby? Thanks.

    Read the article

  • How to use HTTP Live Streaming protocol in iPhone SDk 3.0

    - by Pugal Devan
    Hi Guys, i have developed on IPhone application and submitted to App store. But my application got rejected based on below criteria. Thank you for submitting your yyyyyyyy application. We have reviewed your application and have determined that it cannot be posted to the App Store at this time because it is not using the HTTP Live Streaming protocol to broadcast streaming video. HTTP Live Streaming is required when streaming video feeds over the cellular network, in order to have an optimal user experience and utilize cellular best practices. This protocol automatically determines bandwidth available to users and adjusts the bandwidth appropriately, even as bandwidth streams change. This allows you the flexibility to have as many streams as you like, as long as 64 kbps is set as the baseline feed. In my apps i have to stream prerecorded m4v and mp3 files from my server. I used MPMoviePlayerController to stream and play those videos / audio. How to implement the HTTP Live Streaming Protocol in my apps? Also can i get some sample code? Thanks in advance!

    Read the article

  • Online voice chat: Why client-server model vs. peer-to-peer model?

    - by sstallings
    I am adding online voice chat to a Silverlight app. I've been reviewing current apps, services and SDKs found thru online searches and forums. I'm finding that the majority of these implement a client-server (C/S) model and I'm trying to understand why that model versus a peer-to-peer (PTP) model. To me PTP would be preferable because going direct between peers would be more efficient (fewer IP hops and no processing along the way by a server computer) and no need for a server and its costs and dependencies. I found some products offer the ability to switch from PTP to C/S if the PTP proves insufficient. As I thought more about it, I could see that C/S could be better if there are more than two peers involved in a conversation, then the server (supposedly with more bandwidth) could do a better job of relaying each peers outgoing traffic to the multiple other peers. In C/S many-to-many voice chatting, each peer's upstream broadband (which is where the bottleneck inherently is) would only have to carry each item of voice traffic once, then the server would use its superior bandwidth to relay the message to the multiple other peers. But, in a situation with one-on-one voice chatting it seems that PTP would be best. A server would not reduce each of the two peer's bandwidth requirements and would only add unnecessary overhead, dependency and cost. In one-on-one voice chatting: Am I mistaken on anything above? Would peer-to-peer be best? Would a server provide anything of value that could not be provided by a client-only program? Is there anything else that I should be taking into consideration? And lastly, can you recommend any Silverlight PTP or C/S voice chat products? Thanks in advance for any info.

    Read the article

  • How to set ReceiveBufferSize for UDPClient? or Does it make sense to set? C#

    - by Jack
    Hello all. I am implementing a UDP data transfer thing. I have several questions about UDP buffer. I am using UDPClient to do the UDP send / receive. and my broadband bandwidth is 150KB/s (bytes/s, not bps). I send out a 500B datagram out to 27 hosts 27 hosts send back 10KB datagram back if they receive. So, I should receive 27 responses, right? however, I only get averagely 8 - 12 instead. I then tried to reduce the size of the response down to 500B, yes, I receive all. A thought of mine is that if all 27 hosts send back 10KB response at almost same time, the incoming traffic will be 270KB/s (likely), that exceeds my incoming bandwidth so loss happens. Am I right? But I think even if the incoming traffic exceeds the bandwidth, is the Windows supposed to put the datagram in the buffer and wait for receive? I then suspect that maybe the ReceiveBufferSize of my UdpClient is too small? by default, it is 8092B?? I don't know whether I am all right at these points. Please give me some help.

    Read the article

  • SQL Server 2005 - Enabling both Named Pipes & TCP/IP protocols?

    - by Clinemi
    We have a SQL Server 2005 database, and currently all our users are connecting to the database via the TCP/IP protocol. The SQL Server Configuration Manager allows you to "enable" both Named Pipes, and TCP/IP connections at the same time. Is this a good idea? My question is not whether we should use named pipes instead of TCP/IP, but are there problems associated with enabling both? One of our client's IT guys, says that enabling database communication with both protocols will limit the bandwidth that either protocol can use - to like 50% of the total. I would think that the bandwidth that TCP/IP could use would be directly tied (inversely) to the amount of traffic that Named Pipes (or any of the other types of traffic) were occupying on the network at that moment. However, this IT person is indicating that the fact that we have enabled two protocols on the server, artificially limits the bandwidth that TCP/IP can use. Is this correct? I did Google searches but could not come up with an answer to this question. Any help would be appreciated.

    Read the article

  • Patterns / Solutions to complicated Feature Management

    - by yclian
    Hi all, My company develops CDN / Web-Hosting solution. We have a middleware that's served as a business logic layer and exposes web service for the front-end. I would like to seek for a clean solution to feature management - there're uncertainties and ugly workarounds/solutions in the software that the dev would say "when it happens or is broken, we will fix it". For example, here're the following features that a web publisher can have: Sites limit Bandwidth limit SSL feature + SSL configuration per site If we downgrade a web publisher, when he's having 10 sites, down to 5 sites, we can choose not to suspend the rest of the 5 sites, or we shall prompt for suspension before the downgrade. For the case of bandwidth limit, the downgrade is easy, when the bandwidth check happens, if the publisher has it exceeded, then we will suspend his account. For the case of SSL feature. Every SSL configuration is tied to a site, what shall happen to these configuration object when the SSL feature is downgraded from enabled to disabled? So as you can see, there're many different situations and there are different ways of handling it. I can make a system that examines the impacts and prompts the user to make changes before the downgrade/upgrade. Or a system that ignores the impacts and just upgrade/downgrade. Bad. Or a system designed in a way that the client code need to be aware of the complex feature matrix (or I can expose a helper to the client code to check if a feature is not DEFUNCT) There can be many ways that I am still thinking but puzzled. I am wondering, how would you tackle this issue and is there any recommended patterns or books or software that you think I can refer to? Appreciate your help.

    Read the article

  • Determine asymmetric latencies in a network

    - by BeeOnRope
    Imagine you have many clustered servers, across many hosts, in a heterogeneous network environment, such that the connections between servers may have wildly varying latencies and bandwidth. You want to build a map of the connections between servers my transferring data between them. Of course, this map may become stale over time as the network topology changes - but lets ignore those complexities for now and assume the network is relatively static. Given the latencies between nodes in this host graph, calculating the bandwidth is a relative simply timing exercise. I'm having more difficulty with the latencies - however. To get round-trip time, it is a simple matter of timing a return-trip ping from the local host to a remote host - both timing events (start, stop) occur on the local host. What if I want one-way times under the assumption that the latency is not equal in both directions? Assuming that the clocks on the various hosts are not precisely synchronized (at least that their error is of the the same magnitude as the latencies involved) - how can I calculate the one-way latency? In a related question - is this asymmetric latency (where a link is quicker in direction than the other) common in practice? For what reasons/hardware configurations? Certainly I'm aware of asymmetric bandwidth scenarios, especially on last-mile consumer links such as DSL and Cable, but I'm not so sure about latency. Added: After considering the comment below, the second portion of the question is probably better off on serverfault.

    Read the article

  • Very very slow transfer speeds between Windows 7 and samba server running on Ubuntu 11.10/12.04 minimal

    - by kuzyt
    As mentioned in the title I tried transferring files between Windows 7 and the samba server running on both Ubuntu 11.10 and 12.04 but both showed very slow transfer speeds. Can someone please guide me in the right direction to debug this problem ? wget --output-document=/dev/null http://tokyo1.linode.com/100MB-tokyo.bin --2012-08-21 22:02:17-- http://tokyo1.linode.com/100MB-tokyo.bin Resolving tokyo1.linode.com (tokyo1.linode.com)... 106.187.33.12 Connecting to tokyo1.linode.com (tokyo1.linode.com)|106.187.33.12|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 104857600 (100M) [application/octet-stream] Saving to: `/dev/null' 8% [=============> ] 8,923,980 64.8K/s eta 15m 0s wlan0 IEEE 802.11abgn ESSID:"TNET" Mode:Managed Frequency:2.462 GHz Access Point: 58:6D:8F:26:20:7A Bit Rate=117 Mb/s Tx-Power=20 dBm Retry long limit:7 RTS thr:off Fragment thr:off Power Management:off Link Quality=57/70 Signal level=-53 dBm Rx invalid nwid:0 Rx invalid crypt:0 Rx invalid frag:0 Tx excessive retries:101 Invalid misc:2448 Missed beacon:0 03:00.0 Network controller: Atheros Communications Inc. AR9300 Wireless LAN adaptor (rev 01) Subsystem: Atheros Communications Inc. Device 3112 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx+ Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at fea00000 (64-bit, non-prefetchable) [size=128K] Expansion ROM at fea20000 [disabled] [size=64K] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1+ D2- AuxCurrent=375mA PME(D0+,D1+,D2-,D3hot+,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Enable- Count=1/4 Maskable+ 64bit+ Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [70] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <1us, L1 <8us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <2us, L1 <64us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis+ DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis-, Selectable De-emphasis: -6dB Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap- CGenEn- ChkCap- ChkEn- Capabilities: [140 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- Capabilities: [300 v1] Device Serial Number 00-00-00-00-00-00-00-00 Kernel driver in use: ath9k Kernel modules: ath9k

    Read the article

  • Xubuntu keyboard doesn't work properly

    - by OmPS
    I have installed xubuntu 12.04 but after installation my buttons "d" and "e" doesn't respond properly. When I press "d" on the keypas, it enters automatically giving error as d:command not found, same happenes with Caps lock is on. While presssing "e" i get e+ or +e. Don't know why this weired behaviour, i tired reinstalling xubuntu, but the same think happens again. My keyboard is English(US). Regards,

    Read the article

  • Confusing Callbacks

    - by SullY
    I'm trying to programm now a "game", and started with the EmptyProject that's provided by the DirectX SDK. The problem is that the Callbacks are confusing me. Can please someone explain me? Edit: DXUTSetCallbackD3D9DeviceAcceptable( IsD3D9DeviceAcceptable ); // not sure but I think that's the caps? DXUTSetCallbackD3D9DeviceLost( OnD3D9LostDevice ); DXUTSetCallbackDeviceChanging( ModifyDeviceSettings ); DXUTSetCallbackFrameMove( OnFrameMove );

    Read the article

  • iftop - how to generate text file with its output?

    - by mickula
    iftop is great tool to view almost live bandwidth usage distinguished by source-ip source-port destination-ip destination port. I'm using it to see which client's ip is using most bandwidth. Now I would like to store output somewhere. iftop uses ncurses so iftop > log.txt does not work as expected, result file is not readable. Is there any tool like this which can be used to pipe output to a text file? Thanks for your replies.

    Read the article

  • DD-WRT Connection Leak

    - by Nerdfest
    I have DD-WRT installed on a WRT54G v1.1, and a few of the features seem to cause connections to leak. I've configured it for 1024 connections with TCP/UDP timeouts of 180/30. I've tried higher values as well. Anyway, if I use the Bandwidth tab to monitor the bandwidth usage, the number of connections to my workstation reaches about 450. Is this normal? If not, any idea how to get the connections to either not be created, or to drop much faster?

    Read the article

  • UDP multicast streaming of media content over WIFI

    - by sajad
    I am using vlc to stream media content over wireless network in scenario like this (from content streamer to stream receiver client): The bandwidth of wireless network is 54 Mb/s and UDP stream's required bandwidth is only 4 Mb/s; however there is trouble in receiving media stream and quality of playing specifically in multicast mode; means I can play the stream but it has jitter and does not play smoothly. In uni-cast I can stream up to 5 media streams correctly, but in multicast mode there is problem with streaming just one media! However when I stream from client some multicast streams; the wifi access-point can receive data correctly and I can see the video in "udp streamer" side correctly even when number of multicast streams increases to 9; But as you see I want to stream from streaming server and receive media in client size. Is this a typical problem of streaming real-time contents over wireless networks? Is it necessary to change configurations of my WIFI switch or it is just a software trouble? thank you

    Read the article

  • How to understand the LSI HBA connector specs?

    - by Sandra
    When reading the specifications for the LSi SAS 9206-16e HBA, it says Storage Connectivity; Data Transfer Rates * 16 ports; 6Gb/s SAS 2.1 compliant SAS Bandwidth * Half Duplex 2400MB/s, x4, 6Gb/s SAS lanes Port Configurations * 16 ea, x1 ports (individual drives) * 4 ea, x4 wide ports * 2ea, x8 wide ports Connectors * Four (x4) mini-SAS HD external connectors (SFF8644) So there are 4 physical connectors. Question What is the bandwidth for each of the connectors? I would be temped to say 6Gb/s * 4, but then it mentions the "Port Configurations" and 2ea, 4ea, 16ea, which I don't understand what is. Does this mean, that the 4 physical connectors are not identical?

    Read the article

  • QoS / PBR Routing Questions

    - by Bernard
    I have a 50Mbs Satellite link and a 10Mbs Microwave link supplying a very remote location. Behind these links, I have a 6,400 seat network - with about 3,000 signed in at any one time. My goal is to send all of the Voip traffic (Google Chat, Magic Jack, Skype, Speakeasy, Vonage, Vonage PC, Yahoo) through the microwave link which has 100ms latency. The rest of the traffic can utilize any remaining bandwidth of the microwave link with excess being diverted to the higher latency (600ms) satellite connection. The problem I've had so far is that most automatic routing configurations weigh the bandwidth heavily for preference - and I'm only wanting latency considered. Additionally, I don't know if this can even be handled with the routing hardware I have at my disposal (Cisco 3640, 3745, & 3845). Any recommendations (or really good starting points) would be greatly appreciated.

    Read the article

  • Improving SAS multipath to JBOD performance on Linux

    - by user36825
    Hello all I'm trying to optimize a storage setup on some Sun hardware with Linux. Any thoughts would be greatly appreciated. We have the following hardware: Sun Blade X6270 2* LSISAS1068E SAS controllers 2* Sun J4400 JBODs with 1 TB disks (24 disks per JBOD) Fedora Core 12 2.6.33 release kernel from FC13 (also tried with latest 2.6.31 kernel from FC12, same results) Here's the datasheet for the SAS hardware: http://www.sun.com/storage/storage_networking/hba/sas/PCIe.pdf It's using PCI Express 1.0a, 8x lanes. With a bandwidth of 250 MB/sec per lane, we should be able to do 2000 MB/sec per SAS controller. Each controller can do 3 Gb/sec per port and has two 4 port PHYs. We connect both PHYs from a controller to a JBOD. So between the JBOD and the controller we have 2 PHYs * 4 SAS ports * 3 Gb/sec = 24 Gb/sec of bandwidth, which is more than the PCI Express bandwidth. With write caching enabled and when doing big writes, each disk can sustain about 80 MB/sec (near the start of the disk). With 24 disks, that means we should be able to do 1920 MB/sec per JBOD. multipath { rr_min_io 100 uid 0 path_grouping_policy multibus failback manual path_selector "round-robin 0" rr_weight priorities alias somealias no_path_retry queue mode 0644 gid 0 wwid somewwid } I tried values of 50, 100, 1000 for rr_min_io, but it doesn't seem to make much difference. Along with varying rr_min_io I tried adding some delay between starting the dd's to prevent all of them writing over the same PHY at the same time, but this didn't make any difference, so I think the I/O's are getting properly spread out. According to /proc/interrupts, the SAS controllers are using a "IR-IO-APIC-fasteoi" interrupt scheme. For some reason only core #0 in the machine is handling these interrupts. I can improve performance slightly by assigning a separate core to handle the interrupts for each SAS controller: echo 2 /proc/irq/24/smp_affinity echo 4 /proc/irq/26/smp_affinity Using dd to write to the disk generates "Function call interrupts" (no idea what these are), which are handled by core #4, so I keep other processes off this core too. I run 48 dd's (one for each disk), assigning them to cores not dealing with interrupts like so: taskset -c somecore dd if=/dev/zero of=/dev/mapper/mpathx oflag=direct bs=128M oflag=direct prevents any kind of buffer cache from getting involved. None of my cores seem maxed out. The cores dealing with interrupts are mostly idle and all the other cores are waiting on I/O as one would expect. Cpu0 : 0.0%us, 1.0%sy, 0.0%ni, 91.2%id, 7.5%wa, 0.0%hi, 0.2%si, 0.0%st Cpu1 : 0.0%us, 0.8%sy, 0.0%ni, 93.0%id, 0.2%wa, 0.0%hi, 6.0%si, 0.0%st Cpu2 : 0.0%us, 0.6%sy, 0.0%ni, 94.4%id, 0.1%wa, 0.0%hi, 4.8%si, 0.0%st Cpu3 : 0.0%us, 7.5%sy, 0.0%ni, 36.3%id, 56.1%wa, 0.0%hi, 0.0%si, 0.0%st Cpu4 : 0.0%us, 1.3%sy, 0.0%ni, 85.7%id, 4.9%wa, 0.0%hi, 8.1%si, 0.0%st Cpu5 : 0.1%us, 5.5%sy, 0.0%ni, 36.2%id, 58.3%wa, 0.0%hi, 0.0%si, 0.0%st Cpu6 : 0.0%us, 5.0%sy, 0.0%ni, 36.3%id, 58.7%wa, 0.0%hi, 0.0%si, 0.0%st Cpu7 : 0.0%us, 5.1%sy, 0.0%ni, 36.3%id, 58.5%wa, 0.0%hi, 0.0%si, 0.0%st Cpu8 : 0.1%us, 8.3%sy, 0.0%ni, 27.2%id, 64.4%wa, 0.0%hi, 0.0%si, 0.0%st Cpu9 : 0.1%us, 7.9%sy, 0.0%ni, 36.2%id, 55.8%wa, 0.0%hi, 0.0%si, 0.0%st Cpu10 : 0.0%us, 7.8%sy, 0.0%ni, 36.2%id, 56.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu11 : 0.0%us, 7.3%sy, 0.0%ni, 36.3%id, 56.4%wa, 0.0%hi, 0.0%si, 0.0%st Cpu12 : 0.0%us, 5.6%sy, 0.0%ni, 33.1%id, 61.2%wa, 0.0%hi, 0.0%si, 0.0%st Cpu13 : 0.1%us, 5.3%sy, 0.0%ni, 36.1%id, 58.5%wa, 0.0%hi, 0.0%si, 0.0%st Cpu14 : 0.0%us, 4.9%sy, 0.0%ni, 36.4%id, 58.7%wa, 0.0%hi, 0.0%si, 0.0%st Cpu15 : 0.1%us, 5.4%sy, 0.0%ni, 36.5%id, 58.1%wa, 0.0%hi, 0.0%si, 0.0%st Given all this, the throughput reported by running "dstat 10" is in the range of 2200-2300 MB/sec. Given the math above I would expect something in the range of 2*1920 ~= 3600+ MB/sec. Does anybody have any idea where my missing bandwidth went? Thanks!

    Read the article

  • Configuring and managing Windows web server

    - by Mike C.
    Hello, I run a few websites and I was thinking of paying for a dedicated Windows web server from GoDaddy instead of paying for each site's hosting individually. I know enough about IIS to configure the Host Header and stuff like that, but I'm a little fuzzy about the email portion of the hosting. I have a few questions: Do I need to install an SMTP server on the web server to allow for emails to be sent/received to a website email address? Or is there another approach that I'm unaware of? Are there tools that monitor the amount of bandwidth used by the server? GoDaddy charges for bandwidth and I want to make sure I don't go over. Am I opening a can of worms that I don't really want to open by going the dedicated server route? Things like server updates, security, etc? Thanks!

    Read the article

  • Why do I see a large performance hit with DRBD?

    - by BHS
    I see a much larger performance hit with DRBD than their user manual says I should get. I'm using DRBD 8.3.7 (Fedora 13 RPMs). I've setup a DRBD test and measured throughput of disk and network without DRBD: dd if=/dev/zero of=/data.tmp bs=512M count=1 oflag=direct 536870912 bytes (537 MB) copied, 4.62985 s, 116 MB/s / is a logical volume on the disk I'm testing with, mounted without DRBD iperf: [ 4] 0.0-10.0 sec 1.10 GBytes 941 Mbits/sec According to Throughput overhead expectations, the bottleneck would be whichever is slower, the network or the disk and DRBD should have an overhead of 3%. In my case network and I/O seem to be pretty evenly matched. It sounds like I should be able to get around 100 MB/s. So, with the raw drbd device, I get dd if=/dev/zero of=/dev/drbd2 bs=512M count=1 oflag=direct 536870912 bytes (537 MB) copied, 6.61362 s, 81.2 MB/s which is slower than I would expect. Then, once I format the device with ext4, I get dd if=/dev/zero of=/mnt/data.tmp bs=512M count=1 oflag=direct 536870912 bytes (537 MB) copied, 9.60918 s, 55.9 MB/s This doesn't seem right. There must be some other factor playing into this that I'm not aware of. global_common.conf global { usage-count yes; } common { protocol C; } syncer { al-extents 1801; rate 33M; } data_mirror.res resource data_mirror { device /dev/drbd1; disk /dev/sdb1; meta-disk internal; on cluster1 { address 192.168.33.10:7789; } on cluster2 { address 192.168.33.12:7789; } } For the hardware I have two identical machines: 6 GB RAM Quad core AMD Phenom 3.2Ghz Motherboard SATA controller 7200 RPM 64MB cache 1TB WD drive The network is 1Gb connected via a switch. I know that a direct connection is recommended, but could it make this much of a difference? Edited I just tried monitoring the bandwidth used to try to see what's happening. I used ibmonitor and measured average bandwidth while I ran the dd test 10 times. I got: avg ~450Mbits writing to ext4 avg ~800Mbits writing to raw device It looks like with ext4, drbd is using about half the bandwidth it uses with the raw device so there's a bottleneck that is not the network.

    Read the article

  • How to set WAN side buffers for WRT54GL running Tomato Firmware

    - by Vickash
    I've recently set up a machine running m0n0wall to try and fight buffer bloat and do some traffic shaping. It was more convenient (geographically speaking) to connect the cable modem directly to my old WRT54GL, then pass everything to the m0n0wall machine and have that do the real routing work. It took a bit of work, but it's working pretty well. I have a cable connection. I have m0n0wall set up to utilize only 90% of the specified speed of my subscription, which is fine. But I've noticed that at certain times of the day (possibly when my true bandwidth drops below that 90%), there's more latency if the connection is used heavily, and traffic shaping doesn't seem to work as well. I suspect this is caused by the buffers on the WRT54GL still being unnecessarily large. If the connection is working as expected, they won't get filled, but in times of reduced bandwidth they would. Does anyone know the command I need to execute, on the WRT54GL running Tomato Firmware, to reduce the buffers on the WAN interface to the minimum size possible?

    Read the article

< Previous Page | 16 17 18 19 20 21 22 23 24 25 26 27  | Next Page >