From bandwidth of endpoints

Similar Questions

• How can I limit the bandwidth for endpoint on the router from 1921 to mitigate impacts traffic bittorrent reviews?

  We have implemented an independent network for our wireless comments. It's an ADSL modem, enter a 1921 Cisco. I setup NAT and DHCP on the router and it is only accessible through the wireless connection. We have a connection 50Mbps and about 250 devices connected to it. Some of them use bittorrent and other file-sharing software to quickly eat bandwidth.

  Since the torrent software is adjustable to any port and cannot be blocked like this, I was wondering if there is a way to limit the bandwidth available to each device connection mitigate this.

  I put this in the Group WAN because it seemed appropriate as it performs the external connection. I know that the argument could be made for LAN, because it connects only within the corporate building and not to one of our service centers.

  Any help would be appreciated.

  Hi John,.

  I don't see anything wrong to apply a basic qos strategy, rank traffic using nbar to match with the bittorrent Protocol and other similar protocols. And then just Butterfly traffic down using a map policy.

  HTH

  Mike

• Endpoints registration rejected

  Hi all

  We have a VCS with X6.1 with profile and EX series registered, when I tried that an an iam new getting of registration and in the newspapers of VCS, I could see status rejected for SIP and H323. Reason: "not permitted by the policy".

  Existing end points that were registered before are always saved with H323, but not the SIP.

  IAM attaching snap error log.

  There is no configuration on the registration policy.

  H323 and SIP are on Active watch.

  "15 May 12:28:01 tvcs: event ="The requested record"Service ="SIP"Src - ip ="200.101.2.32"Src-port ="5061"Protocol ="TLS"AOR ="[email protected] / * /"Contact =" sip:[email protected]/ * /: 5061; transport\ = tls"duration ="60"Level = '1' elements UTCTime ="2012-05-15 08:28:01, 285"

  "15 May 12:28:01 tvcs: event ="Update record"rejected reason ="AOR is not authorized by the registration policy"Service ="SIP"Src - ip ="200.101.2.32"Src-port ="5061"Protocol ="TLS"AOR ="[email protected] / * /"Contact =" sip:[email protected]/ * /: 5061; transport\ = tls"duration ="60"Level = '1' elements UTCTime ="2012-05-15 08:28:01, 285"

  "15 May 12:27:58 tvcs: event ="Rejected registration"reason ="Not permitted by the policy"Service ="SIP"Src - ip ="200.101.70.11"Src-port ="5061"Protocol ="TLS"AOR ="[email protected] / * /"Contact =" sip:[email protected]/ * /: 5061; transport\ = tls"duration ="60"Level = '1' elements UTCTime ="2012-05-15 08:27:58, 897"

  "15 May 12:27:58 tvcs: event ="The requested record"Service ="SIP"Src - ip ="200.101.70.11"Src-port ="5061"Protocol ="TLS"AOR ="[email protected] / * /"Contact =" sip:[email protected]/ * /: 5061; transport\ = tls"duration ="60"Level = '1' elements UTCTime ="2012-05-15 08:27:58, 897"

  "15 May 12:27:58 tvcs: event ="Update record"rejected reason ="AOR is not authorized by the registration policy"Service ="SIP"Src - ip ="200.101.70.11"Src-port ="5061"Protocol ="TLS"AOR ="[email protected] / * /"Contact =" sip:[email protected]/ * /: 5061; transport\ = tls"duration ="60"Level = '1' elements UTCTime ="2012-05-15 08:27:58, 897

  No matter who came across this?

  So if [email protected] / * / is already registered on SIP, is the registration request for [email protected] / * / of

  200.101.2.32 from another point endpoint which currently holds this inscription?

  Is CPL/Admin Policy active on the VCS?

  Initially, you mentioned that no registration policy was in use, but can you please check that VCS Configuration > registration > Configuration > Restriction policy is set to "None"?

  -Andreas

• Get a variable from an open tool

  Hello
  
  I m developed a tool open in Java to get an ID of a HttpRequest.
  I want to keep this answer (a string) in a variable for use in another step.
  
  Is this possible?
  
  So, I tried to create a .jar I used in a procedure like this thanks to http://askankit.blogspot.com/2010/09/call-java-jar-methods-from-odi.html:
  Import os
  import sys
  
  jars =]
  'F:\NetBeansProjects\UrlEndPoint\dist\UrlEndPoint.jar '.
  ]
  
  for pot in pot:
  sys. Path.Append (jar)
  
  from com.mycompany.endpoint import *.
  
  FW = urlendpoint)
  FW. Login (URL, USER, PASSWORD)
  
  But the operator returns this error:
  org.apache.bsf.BSFException: exception of Jython:
  Traceback (innermost last):
  "< String >" file, line 13, inside?
  java.lang.NullPointerException
  
  at java.lang.Class.isAssignableFrom (Native Method)
  
  at org.python.core.PyJavaClass.init__class__ (PyJavaClass.java)
  
  Someone knows how to solve this kind of problem?
  
  Best regards
  Julien
  
  Published by: user12115064 on October 1, 2010 05:28

  In the procedure of ODI.
  As a first step, use
  Command on target - as the tank of Java Bean technology
  <>
  You must write your java code this symbol
  @>

  In the next step, provide the necessary technology

  call the variable as <@ variable="" @="">as shown in the first response.

  You can also find another thread that should give you a better idea.

• When a character appeared reading serial port

  How to wait for some specific characters occurred in the serial port (e.g. port COM1 RS232 on PC) and then they recover at the port?

  I want to communicate back with a motor controller that uses ASCII strings such as commands and responses. It formulates a response to any command sent, and the response contains exactly a termination character (that I can specify during installation) at the end of the response string. Sometimes also, it sends a message when there is no order issued, for example a disc error message. There is no simple way and reliable when the controller is going to speak, when he won't, and the message will be exactly how many time, but we do know that each message will have this stop only at the end character. I would like to interpret the entire message in my code, that is to say, I would like to retrieve the string of all the characters from the previous endpoint character up to and including the most recent stop character. I think it means that I would have a VI that returns the message string and does not stream until the stop character appeared and was added to the response string. Or, Alternatively, a loop that adds entire messages to a queue of strings.

  All the screw example I found seems to rely on a certain number of milliseconds to wait or to know how many bytes to read, in order to use VISA Read.

  So far, I use a loop which seeks bytes in the buffer, retrieves everything to add to a string of shift register and test if there is a character of the string endpoint, all extract up to and including the stop character, if so. This feels very awkward and expensive for what should be a common task. Is this general law approach, or did I miss something in a simpler way?

  I read on the communication by Message and characters of endpoint, approach that sounds functionally similar to this, but it seems around standards of SCPI and my motor controller does not support this. In any case example Finder does not get a single hit on 'Message '.

  Thank you!!

  It really looks like you are doing things a lot more difficult it must be. Look at the VISA configure Serial Port. It has a character of endpoints allow and end characters entries. If you wire a real (or leave that he unwired) entry activate, read VISA will end automatically when you specify the stop character is detected. As long as the number of bytes to read is larger that the largest string that you expect to read, there is nothing else you need to do. That's how examples of shipment are put in place and discussions about the characters of the termination. Should there be nothing related to sustainable intensification of CROPS. If you do not get a message in your specified time-out, you get a time-out error.

• Jabberguest delay compensation appeal

  Hi all

  I am currently using Jabberguest 10.6.8.11.  I discovered a problem with the dial from mobile devices, endpoints.  If the application is closed without hanging up the call, or if there is a loss of network on the mobile device, then the call clears out.  If the user reloads the app or retrieves the network connectivity and calls once again, the system will always be in a call to the original session.

  Does anyone know of a way to reduce the time-out of the call session when no traffic is received or not end the call when the server loses the connection to the client remote etc?  I have reduced the VCS SIP session refresh interval, but it is still 90 seconds before the call is disabled.

  See you soon.

  Peter.

  Hi Peter,.

  10.6.10 JabberGuest has provided an option set in comment server Jabber web admin to set the call session expires time (default is 60 seconds), please refer to Jabber comments 10.6.10 release notes

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/Jabber/guest/10_6_10/...

  Concerning

  Yu

• What is the correct Version of software E20

  Hello.  I have an E20 which I am trying to set up.  After recording combined as 'root' via the cable from the console connected to the port, I have a problem.  When I entered the IP address information via the touchpad and the restart, he writes nothing to the unit.  When I am logged in as root to check, I see that I don't always have an IP address that is assigned to eth0.  When I hard code one in through 'root' for eth0, rises the interface and it works.  Then and only then I can web browse to the E20.

  I use the version of the TE4.1.1.273710 software which is what it came with.  Y at - it a version more recent and if so, please give the link.  I have not any other images stored on the device.

  The only way I could get it back by default with success has been to use this command from (date) "telepresence Endpoint technology Handbook": rm mnt/base/active/config.db.

  Note: the correct syntax should be rm /mnt/base/active/configuration/config.db

  You should should not unzip the pkg file, just download this file in the e20 through its Web page, or whatever your usual method for upgrades.

  Thank you
  Guy

  Sent by Cisco Support technique iPhone App

• Video calls to Highway B2B - license Question

  I have a question about licensing Expressway C & E calls video numbering URI of B2B (via internet).

  You will need a license from RMS on Hwy C & E if your codec (SX20) is registered to your CUCM when do B2B video URI calls?

  If you make a call from a registered endpoint CUCM point and the other end is not on your call Manager (ie: B2B), you will need licenses RMS.

  If you make a call between two ends of CUCM, where it is written internally to call the Manager and the other is registered via Mobile and remote access, you don't require licenses RMS because end points recorded for call manager using MRA do not consume a license of appeal.

• Output of "crypto ipsec to show her.

  Hello

  In a VPN l2l baseline using ezVPN, the server behind the NAT, client device using 3 G. What would be the reason to have the output of the show crypto ipsec sa, a current peer differs from crypto remote endpoint on the server?

  Thanks for your help.

  David

  Given that it is behind a NAT and NAT device is only the layer 3, and it changes the contents of the IPSec VPN when negotiating because it is encrypted, so you might see counterpart current differs from endpoint remote crypto on the server.

• GW ISDN Cisco with VCS control

  Hi Experts,

  I'm new with Cisco ISDN Gateway 3200, and I will integrate to the existing video network managed by VCS control with the endpoints and MCUS registered as SIP.

  Here is an example of the existing video network numbering plan:

  SIP URI endpoint: [email protected] / * /

  MCU SIP URI: [email protected] / * /

  ISDN GW H323 alias: 54xxxxx

  I came up with these questions:

  1. for us intergate ISDN GW on the video network, we must save as H323 on the right VCS?

  2. to call scenario as a point endpoint SIP or MCU calling to one ISDN endpoint via video, how the call flow? Make the registered endpoint/MCU SIP point as SIP on the VCS can dial directly on this endpoint ISDN? Or they will call first the H323 number of ISDN GW recorded on the VCS then routed to an auto attendant of the ISDN GW?

  3. How about ISDN endpoint SIP endpoint?

  Please send me a sample of guide and the configuration of the ISDN GW and VCS about how we could improve the flow of calls work on the call scenarios mentioned.

  SIP---> ISDN

  ISDN---> SIP

  Thank you very much for the help.

  Best regards

  Acevirgil

  Hello

  1. For us to intergate the ISDN GW on the video network, we need to register it as H323 on the VCS right?
  

  Yes, it's true. This isn't the only method, but it is the most used and most easy way. I suggest you use it.

  2. For call scenario like from a SIP endpoint or MCU calling an ISDN endpoint via video, how's the call flow? Do the SIP endpoint/MCU registered as SIP on the VCS can dial directly on that ISDN endpoint? Or they will dial first the H323 number of the ISDN GW registered on the VCS then routed to an auto attendant of the ISDN GW?
  

  In this case, you will need to ensure interoperability the call in VCS. No matter what SIP endpoint can dial numbers ISDN, and then VCS will route the call to the gateway h323 format, only the number with any ' @domain.com ' and the call will be interoperability. The flow would be something like this:

  Point endpoint [sip] SIP----> - VCS [H323]---> Gateway ISDN

  3. How about from ISDN endpoint to SIP endpoint?
  

  The same concept is applied. ISDN gateway sends the call in H323 for VCS and interwoks VCS the call and sending SIP endpoint. Something like this:

  SIP endpoint<-------[sip]--------- vcs=""><--------[H323]---------- gateway="">

  With regard to incoming calls from gateway ISDN to VCS, there is one important thing to consider. You basically have to methods:

  • You can configure the ISDN gateway to route calls to the auto attendant, then users will be able to recompose the verse numbers of internal endpoint for the auto attendant
  • You can configure the ISDN gateway to use DID (Direct inward dialing), in this case, you create routes to the gateway which maps each ISDN number to an internal endpoint registered to the SCV

  Both methods work fine, however, when using auto attendant, it is very important to implement a scheme of toll-free fraud prevention. Take a look at this thread:

  https://supportforums.Cisco.com/message/3971947#3971947

  Regarding the guide, it is not a simple step by step explaining, but the Administrator's guide provides a good explanation on how to configure the dial plan in the gateway and how to enter the doors of VCS. Check out this guide:

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/isdn_gw/admin_guide/isdn_gateway_printable_help_2-2.PDF

  I hope this helps.

  Concerning

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• VCSC INACCESSIBLE DESTINATION

  ALCON,

  I have a neighbor relationship upward with a site and the site remote is able to dial in our bridge via IP and alias.  I am able to dial up a point to end of line remote via alias.

  Problem is when the line trys remote dial in one end on my end which is recorded in the cluster, it does not connect.

  Don't have not even get a ring.

  The system is functional at 100%.  I can dial the far end via alias no problem...

  I checked the search history and see that the system is trying to connect and an error message "Destination unreachable"? So it appears the call makes my VCS cluster but will route to the endpoint?

  Any ideas.

  I can make other calls in two ways by nearby with other sites of relationship alias.

  Thank you.

  Chet Cronin
  801-815-3539 (USA)
  + 9379 601 - 3954 (Afghanistan)

  Hello

  If your VCS is running in Direct mode, theoretically, you don't need to create a rule to search to any IP address.

  Tell me something, have you tried to call this end point of 1000 MXP numbering by IP address from a local endpoint registered to the same local VCS? It work?

  Also, you can try temporarily disabling the search rule 'any address IP"and try to call again to the remote site? You can post the details of the history of the research?

  In addition, if you try to call this end point the remote site MXP component number instead of the IP address, it works? Can you confirm?

  Concerning

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• Capture OCAP Codian

  Hello world.

  Can someone help me please. I would like to play a video call from Pcap file. I heard that we can seize wireshaek catches on Codian MCU

  and play the video file to check the quality of the video for the loss of package lagg etc.

  Please suggest.

  Thank you

  Video

  Just found out today through a case of TAC, you can do packet capture directly from the MCU, there are two warning however and are noted near the bottom.

   True for MCU 4.2 or newer, and Telepresence Server In order to start the capture, you will need access to the console port of the MCU/TS. MCU:> nettap usage: nettap [-a|-l|-s|-h] A|B -a : capture all packets (i.e. disable most of filter) -l : disable limit on number of packets captured (160000) stop with Ctrl-C -s : disable 128 byte limit on packet length -h  : only capture packets to from  § The A | B refers to port A or port B. In almost all cases you will want port A. For example, if you want to capture media coming from a particular endpoint at 192.168.0.5, you would use: MCU:> nettap -as -h 192.168.0.5 A Don't forget the -s, or you will only capture the first 128 bytes of each packet - no good for media (and not much good for protocol signalling either). The capture can be retrieved from the MCU/TS using the Web interface: Status > General > Download network trace. It's a good idea to delete it after downloading if CDR logging or Audit logging is enabled. Warnings Using on a busy MCU will cause problems Processing power is limited on Codian products, especially for 4200/8420, IP VCR and ISDN GW. Using nettap on a busy MCU is a lot of work (the MCU will be dealing with a LOT of traffic), and this could cause performance issues and potentially even stability problems. You will run out of space Space is also limited on Codian products, so capturing media for an extended period of time is not an option. Leaving a large trace on the box will also severely limit the space left for Audit and CDR logs.

• Issue of outside calling on Cisco MX200G2

  Hi all

  I have MX200G2 registered on CUCM. When I compose the other video settings registered to CUCM call works fine.

  Now if I want to call anyone outside my organization IE I want to compose a public IP call does not pass through. Anyone can guide me please how to do the config for the same thing?

  My CUCM version: 10.5 (2)

  Screenshot attached to the call failed:

  Disconnect the cause	Bad request - 'URL incorrect or absent.
  Disconnect reason code	400 (SIP)
  Disconnect the lead case	RemoteDisconnect
  Type of event	NoAnswer

  Thank you

  Just to be safe, you attempt to dial the SIP URI or IP address?  In addition, your screenshot is not downloaded/attached to your message.

  If you want to dial an IP, CUCM does not support this out of the box, but you can make it possible that by using your VCS-control/channel Express or Freeway-Center/periphery, see the document below.

  Dial IP addresses from items of endpoint CUCM VCS / Expressway Configuration example

• Manual vs Auto Transitions

  If I want to automate the application of default transitions between many clips in a sequence, I can highlight clips and drag them to the icon to automate to sequence, make my transition selections (overlay edit) and it's done.  This process will apply transitions on unpublished original clips without sleeves and without repeating frames.  Thus, the automatic process of the sequence must create sleeves (who were not part of the original item) and apply transitions without repeating frames.   However, if I manually insert the same unpublished original clips without handles in a sequence, select all the items, and then choose sequence > Transitions apply to selection... the process will NOT create the handles needed to apply transitions without repeating frames.  Transitions by default will apply but WITH repeated frames...

  Is there a method to implement the In and handles advance unreleased clips that could automate the manually the method of application of transitions?  I assume repetitive frameworks are bad and it's pretty painful and long to manually create handles at the end and the beginning of many executives before you manually apply a transition...   I am using PP CS5.5 and work with AVCHD m2ts files.   Thank you

  Greg

  Just overlap the previous clip by the number of images you want for handles.  So if you need a transition of 30 images, then drop the 30 images new clip from of the endpoint of the previous item.

  Tip: If no clipping is selected in the timeline, typing in a number like 30 on the pad moves the playhead to this number of images.  Similarly, a 30 entry on the keypad goes back the playhead to this number of images.

  -Jeff

• Bandwidth dedicated to endpoints

  Hi all

  I have a question:-How can I refrain from provision of bandwidth dedicated to my endpoints in the entire network infrastructure, for example, call HD needs minimum 2 Mbps for a good call. So, if I have provided dedicated 2 Mbps of bandwidth to the endpoint, there is no need that I'm reserving bandwidth for an endpoint continuously.

  Is there anyway where I allocate bandwidth of an endpoint in an ad hoc way, where and when necessary so that it is not bandwidth reserves at all times. ?

  Look forward to your valuable suggestions.

  Kind regards

  Saurabh Gupta

  Saurabh, your endpoints not use bandwidth when they are not in a call.  There may be a very small amount of traffic RAS, but it of very small and go unnoticed by the other devices on your network.

  Usually most of the customers either do not worry about the reservation of bandwidth for video, or they use QoS mechanisms on their network to ensure that video (which is real-time and sensitive to delays and loss) is not affected by other data needs.

  This is different from the bandwidth dedicated , where the bw is set aside plenty of time for only video.  QoS comes into play to make sure that the video has the bw that he needs to start your network at congested.

  I've personally never seen anyone try to devote a part of their network bw exclusively for the video.  If they go that far, they usually have a completely separate network for video.