Question of MCU 4505

Similar Questions

• Software update of MCU 4505

  To get back into the mix of things with our telepresence equipment and we have a MCU which is on an older version of software.  After downloading the new software MCU 4.5, I do just that if I update I will not lose one of my features enabled.  Our current version is 4.2 (1.46) version of software.  We have the following features enabled:

  1. Encryption
  2. Web conferences
  3. Activation of MCU 4505
  4. key feature of capacity 1080 p upgrade

  Looking at the docs online, doesn't look like I need anything that either step upgrade.  Already upgrades for my C40s since these option buttons do not disappear when you upgrade to a newer version, but was not sure of the MCU.  Any ideas would be great.

  Hi, Brandon

  Options keys do not disappear.

  Your current options keys remain valid with the new MCU software.

  BR Oleksandr

• Codian MCU 4505 content encryption is not possible

  Hello

  I'm having a problem with MCU 4505:

  If I put the SIP trunk to the Codian MCU (to route calls with a specific prefix) for the MCU to TLS instead of TCP, the content channel could not be seen on other participants. I see in the statistics of participants, that the content string is sent from one endpoint to the MCU, but not received from the other particpiants. In the view of the Conference, I saw the following message:

  TX of content: encryption is not possible

  I checked the docs, but the description of this cause is not really free explaining:

  The MCU is unable to send the encrypted video content to this participant.

  If I put the trunk for TCP, no encryption is used and the sharing prentation works as expected. Basically, the encrpytion works fine, because the audio and video channels are received and sent as expected.

  No matter who else knows this problem or ideas how to solve this problem?

  BTW: Versions of software:

  MCU 4505 4.3 (1.68)

  EX60 TC 5.1.4

  Jabber 4.5

  C20 TC 5.1.4

  

  Kind regards

  Paul

  It is noted as a known bound in notes;

  "The transmission of content SIP of the MCU by using the binary protocol (BFCP) ground control is not supported on.

  encrypted calls. To allow content to pass on SIP calls in a separate main video channel, you should

  Disable encryption on the MCU or target endpoint. »

  The MCU supports the transmission of content encrypted on H.323 calls only, but not the SIP.

• Allowing endpoints make multi-point on Cisco 4505 MCU calls

  Hello

  I have a Cisco 4505 MCU and several endpoints.  Is there a way to allow that my end point called multipoint adhoc using MCU 4505?  For example, I would say that dial site1 using mcu, then dial site2 help mcu or call a group (site 1 and site1...) using mcu and it would know that it is supposed to use the 4505.

  Thank you

  Ravi

  Of course, you can configure (a few 100) static, meeting rooms, auto create conferences on the fly, based on a model or have the means put in place and there is even an on screen menu allows users to create or join conferences.

  Check out the example (page 37ff) Administrator's guide

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/MCU/admin_guide/Cisco_TelePresence_MCU_4-4_Product_administration_guide.PDF

• CUCM (Cisco Jabber) - VCS - MCU (failed appeal of SIP - SIP)

  Hi all

  in our Videoinfrastructure, there is a little problem with cisco jabber client registry on the CUCM to our MCU.

  The MCU is inscribed on the VCS.

  If I make a call my Client of Jabber Cisco SIP call has failed. I call the number 4925488XXX. The callingparty is the 25203232.

  In the CUCM, there is a Routepattern who translated the «Called Party Number» number = 8099 (this is the SIP Trunk for VCS).

  Called Party Transformations:

  Called mask = 8XXX

  Take the numbers Instruction = None

  Prefix =

  Called number = 8099

  In venture capital, there is a transformation for the 8099 dialed.

  

  The MCU is registerd with [email protected] / * / on the VCS - it's self attandant.

  In the Cisco Jabber client I never receive video or audio of the MCU AA - it looks like the call's not connect.

  If I spent a glance in the VCS, I see the appeal as active and connected.

  

  Also, I traced the call made by the VCS in debug mode.

  SIPMSG:

  | SIP/2.0 180 ringing

  Via: SIP/2.0/TLS CUCM IP SIP; direction = z9hG4bK10ea4511f7965; has received = CUCM; penetration area = CUCMUC

  Call ID: [email protected]/ * /.

  CSeq: INVITE 101

  Contact: <> [email protected] / * /; GR = urn: uuid:c4a744d7-5975-52db-b21c-8c519fbab06e >; isfocus

  From: Stefan Walter "' <> [email protected]/ * */>;tag=252394~c75f2de1-fd7b-43c2-8bd0-6b72860c9048-46890229

  To: <> [email protected] / * / IP >. tag = 11B05829C2080000

  Record-Route: <> [email protected] / * / SIP; transport = tls; LR >

  Record-Route: <> [email protected] / * / SIP; transport = tls; LR >

  User-Agent: Codian MCU 4505 v4.3 (2.18)

  Content-Length: 0

  |

  "2012 10-26 T 15: 33:25 + 02:00 DDUCP001 tvcs: elements UTCTime ="2012-10-26 13:33:25, 230"Module =" network.sip"Level ="INFO": Src - ip = 'CUCM IP' Src-port ="39813"detail =" receive the method Request = CANCEL, Request-URI = sip:[email protected] / * / IP, Call-ID =[email protected] / * / IP.

  2012 10-26 T 15: 33:25 + 02:00 DDUCP001 tvcs: elements UTCTime = "2012-10-26 13:33:25, 230" Module ="network.sip" Level = "DEBUG": Src - ip = 'CUCM IP' Src-port "39813" = "

  SIPMSG:

  | CANCEL sip:[email protected]/ * /: 5061 SIP/2.0

  Via: SIP/2.0/TLS CUCM IP SIP; direction = z9hG4bK10ea4511f7965; receipts = 10.202.131.96

  Call ID: [email protected] / * / IP

  CSeq: the 101 CANCEL

  From: Stefan Walter "' <> [email protected] / * / IP>;tag=252394~c75f2de1-fd7b-43c2-8bd0-6b72860c9048-46890229

  To: <> [email protected] / * / IP >

  Max-Forwards: 70

  Date: Friday, October 26, 2012 13:33:25 GMT

  Reason: Q.850; cause = 3

  Content-Length: 0

  ########################

  Reason: Q.850; cause = 3; text = "no route to the destination.

  Do you have any hint for me where I can troubleshoot further? It's a question MCU or CUCM or VCS?

  Good bye

  Stefan

  Hi Stefan.  I would probably run a Trace SDI on the CUCM as well.  Seems the cancel is coming from CUCM side and end the call.  To check that your possibly fall on: CSCty07061, I could run the trace of the SDI and see after reception of 180 ringing of venture capital if he tries to do a DNS query for @mcu.

  The entries may look like:

  | 1.3508793, 63, 100, 2 ^ 10.52.36.15 ^ *.

  11:20:09.377 | / / SIP/SIPDns (2,72,1) / wait_SdlDnsSrvRecordRsp: received specified SdlDnsSrvRecordRsp is - 1. 0, 0, 0, 0.0 ^ * ^ *

  11:20:09.377 | / / SIP/SIPDns (2,72,1) / copyRecordRspToRecordReq: query retry SRV as a 1 request | 0, 0, 0, 0.0 ^ * ^ *

  11:20:09.377 | / / SIP/SIPDns (2,72,1) / wait_SdlDnsSrvRecordRsp: hostname (DNS A or AAAA query called a query SRV Fail) = mcu

  --------------

  If that's what your race, false address to solve the @mcu would be required for the call to go forward.  I have this based off the coast an explanation here, but cannot actually ensure that it's what your run in.  But check the trace of the SDI AFTER the 180 ringing and see.

  I hope this helps.

  VR

  Patrick

• Join the public network through the firewall option MCU Conference

  Hello world! Lately, we have a goal to jabber (without vpn) of internet users to join conferences within our network. Our telepresence infrastructure includes VCS, MCUS and endpoints (MX, SX). I discovered that MCU (4505) has a feature called video firewall option (under license). My question is whether it will be sufficient to achieve our goal? I've also heard that VCS expressway is used. What is the difference between these two?

  Firewall MCU option just the second NETWORK card, so you can have one with a public IP address and the second with a private IP address (essentially NAT'ed), and the internal and external clients will be connecting to a virtual meeting room common pre-configured on the MCU.

  The VCS-E (Expressway) works in tandem with your VCS - C to offer secure firewall courses. VCS - C is located on the inside of you firewall and the VCS-E is normally in the demilitarized zone.

  This allows, among other things, external systems to call your systems using H.323 and SIP URI dial, this includes clients of JabberVideo, but not Jabber client CPU.

  For Jabber clients you would deploy normally expressway-e series, see http://www.cisco.com/c/en/us/products/collateral/unified-communications/expressway-series/datasheet-c78-730478.html and http://ciscocollab.wordpress.com/2014/01/29/deploying-collaboration-edge/ - whether it will work with your existing deployment is another story, you might be better off ultimately to something similar to Pexip or Acano - both were developed by former TANDBERG/Codian/Cisco engineers.

  /Jens

  Please note the answers and score the questions as "answered" as appropriate.

• Cisco TelePresence MCU disconnects a session IP

  Now I have a MCU 4220, two VCS (neighbouring) and a version endpoint MXP 95. MCU is 4.3, and the version of VCS is X5.2.MCU are recorded in a vcs and 95 MXP is recorded in other vcs. Two VCS are nearby. Now MCU makes a call with MXP 95, but after a minute or so, MCU drops the call. I tried several times, and the situation is the same. The message to MCU I founded is "h.245 network connection error". I checked the network, everything is OK. I took an another MCU 4505 to test. Nothing wrong happene and the call can work well.

  Please give me some advice.

  Thank you.

  Hi do you have a firewall on your network where traffic can be? See you on your video mxp for this minute?

  / Magnus

  Sent by Cisco Support technique iPhone App

• What is the best practice? Several MCU Multisite conferences

  Hello

  We currently have two independent sites, each with their own MCU (MCU 8500 and a MCU 4505), several points of endpoint and VCS ExpressWay infrastructure management (TMS). Currently, we are creating a conference in each site that has the MCU on the other site as a participant. It's not ideal because we get several images in the image, especially when you use a format of 7 + 1 split for each MCU.

  What is the ideal way to handle this? The options I can think of are:

  1. Have every endpoint call an MCU instead to use both.
  2. Use a format full screen instead of 7 + 1 split.
  3. Place all the VC infrastructure in to a TMS win something?

  Thank you

  Mark

  Hi Mark.

  1. ask to all the endpoints call an MCU instead to use both.

  This will fix the problem of page layout that you face and you can use all the features of conference as control of floor etc.

  Disadvantage will be the consumption of bandwidth, every EP of another site will consume call bandwidth, if 5 Ep calling another MCU @ 1 mb, then you will need 5 MB for confrance.

  If you have enough bandwidth it will be the right option.

  2 use a format full screen instead of 7 + 1 split.

  Advantage for this config is 1 MB to call far you need to finish MCU form local MCU.

  Disadvantage - layout.

  3. put all the VC infrastructure in the TMS win something?

  Nothing.

  HTH

  Kind regards

  Dharmmesh

• Two mcu 4205 cluster?

  Two mcu 4205 (4.4) with the same software both have 12 media port, now I have to make them work together as an mcu and 12 multimedia ports, have also the active guard and can be composed of devices or conferences by identification number, also I need if one of them is toward the bottom of the can function as main mcu and conferences continue

  How can I get there?

  Grouping/stacking of the MCU is not supported for 4200 series or 4500 to two MCU operate together as one.

  Clustering/stacking is supported only for the 5300 MCU series. You can only cluster/stack two units of the MCU.

  The idea of clustering/stacking the MCU is to increase capacity. The cluster will provide the sum of the capacities of the individual units. For example, if group you two microcontrollers that each supports 10 ports x 1080 p you supported a cluster which takes 20 ports x 1080 p.

  Optionally you can mount your two cascading 4205 microcontrollers:

  http://www.Cisco.com/c/en/us/support/docs/conferencing/Telepresence-MCU-4505/118763-config-MCU-00.html

  Another solution is that you need a Cisco conductor who works with VCS/CUCM.

  Take a look at these links for more information about deploying Cisco conductor:

  http://www.Cisco.com/c/en/us/products/collateral/conferencing/Telepresence-conductor/data_sheet_c78-709046.html?CacheMode=refresh

  http://www.Cisco.com/c/dam/en/us/TD/docs/Telepresence/infrastructure/conductor/config_guide/XC3-0_docs/Telepresence-conductor-Cisco-VCs-B2BUA-deployment-guide-XC3-0.PDF

  http://www.Cisco.com/c/dam/en/us/TD/docs/Telepresence/infrastructure/conductor/config_guide/XC3-0_docs/Telepresence-conductor-unified-cm-deployment-guide-XC3-0.PDF

  Kind regards

  Acevirgil

• Conductor XC2.2 and TMS 14.3 automatic connects

  Hello

  I've also seen a topic on current versions of beta XC2.2 conductor and TMS 14.3, so I'll post my question here.

  I use the conductor with VCS, CUCM, MCU 4505 and TS. I am able to create ad hoc via CUCM and VCS and regular calls through TMS conferences.

  If I create a conference with automatic connect and the conductor in the call I can see that the MCU or TS tries to achieve a participant, but not with the URI. To use my email address (findme ID) to call me, but instead, it uses strange URI identifiers: [email protected]/ * /: 5073

  1.1.1.1 is the IP address of the driver. On the front part @ I do not know where it comes from, but it looks like an internal identifier.

  These are the logs of the MCU for a test with a participant conference to be connected automatically:

  13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: "Videokonferenz 101" is 13307 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101], numerische Favorites auf ID '88800010 '. 13308 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101], pine Favorites auf "114". 13309 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101], Gast-PIN options auf "114". 13310 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [101 Videokonferenz], busy 'public '. 13311 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101] content mode set to 'convert '. 13312 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101] Stummschaltung EIB participation [Audio = deaktiviert] 13313 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101] Stummschaltung EIB participation [video = deaktiviert] 13314 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [101 Videokonferenz], Layoutsteuerung uber FECC/DTMF "use FECC and relief to DTMF". 13315 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101], maximum Dauer Favorites auf 'permanent '. 13316 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101], Ende Favorites auf "kein Enddatum. 13317 13:38:05 API admin /HTTP/TCP/1.1.1.1:38141 Info conferences: Conference [Videokonferenz 101] auto remove time limit set to "600 seconds". 13318 13:38:06 admin API /HTTP/TCP/1.1.1.1:38141 API-Endpunkt-Einstellungen Info: Endpunkt [e922d880ee1854c242ba94e69b113af: [email protected]/ * /: 5073] via API zu Conference "Videokonferenz 101" added 13319 13:38:06 admin API /HTTP/TCP/1.1.1.1:38141 Info API-Endpunkt-Einstellungen: Endpunkt [3805: [email protected]/ * /: 5073] via API Conference "Videokonferenz 101" added zu 13320 13:38:36 API admin /HTTP/TCP/1.1.1.1:38149 Info API-Endpunkt-Einstellungen: Endpunkt [3806: [email protected]/ * /: 5073] via API zu Conference "Videokonferenz 101" added

  Unfortunatetly newspapers are in German, but I think you can see that there are a few strange added URI.

  If I create a conference without a conductor the log shows the URI:

  13386 14:07:14 /HTTP/TCP/172.21.10.12:52001 Info API-Endpunkt-Einstellungen admin API: Endpunkt [Paul Woelfel: [email protected] / * /] via API zu Conference ' 1234 - Videokonferenz 07/08/2013 "added.

  1.1.1.1 is the IP of the conductor (of course not the real IP ;-))

  Someone at - test the latest versions as expirenced this question?

  Hi Paul,.

  You use the conductor and VCS B2BUA deployment? If so, this behavior is correct. MCU must not call endpoint directly, MCU must compose a unique URI of the driver, something like [email protected]/ * /, so Condcutor will take this URI and route the call to VCS replacing this strange URI of the right destination SIP URI according to what you have planned.

  But to make it work, MCU must not be registered for VCS, MCU should only be integrated with driver, without registration, SIP and H323 for VCS.

  Can you confirm what type of deployment you have? What integration between your MCU and conductor?

  Concerning

  Paulo Souza

  Please note the answers and mark it as "answered" as appropriate.

• Tandberg Multi Point called endpoints

  Hello Experts from Cisco TP,

  We have Tandberg Profile and Ex 90 end points, I understand the EX series and profile series integrated MCU resource, how do I configure Multipoint calls without using a separate MCU.

  We have ver.6 VCS and MCU 4505. Can make Conference calls using MCU 4505.

  Thank you

  Bobby

  Hi Bobby,.

  You can foind installed on endpoints, it is visible on a GUI on the system information page, section Installed options

  According to the description of the product of C90, it contatin multisite functionality, if it isn't on EX90, then perhaps he is not bought for it.

  You can check the license using the serial number of the device and licensing tool

  Link to the tool: https://tools.cisco.com/SWIFT/LicensingUI/Home#tabbeditem

  Reach:

  • Tandberg Software Upgrade & search

  then click on search and return and step

  Kind regards

  Vivek

• The IP address private VC directly to the public IP address

  Hello

  I'm a bit puzzled as to why a specific call, I saw worked for a couple of guests and wonder if there was no change in the situation in the H.323 protocol that allows a form any NAT crossing built natively into the codec without involving and external gateway function.

  the reason I ask is the following

  I got a call from a customer with a codec on a private no routable IP to my system that is located on a public IP address, the client had no details of NAT configuration in the endpoint and was able to call my system directly without issue by calling directly to my public IP address.

  historically now if I had a system on a private IP address was sitting behind a NAT, I expect that the public system IP would see no routable IP address of the H.225 message and try to answer the private IP RTP media that would not go through, it does not seem to occur.

  the call that I have lived seemed ends without problem, media flowed in both directions.

  My endpoint is a Cisco edge 85 on the version of the firmware F9.x

  the other codec parts is an 85 edge on the version of the firmware F9.x

  My codec is on a public IP address that is completely open to the H.323 ports

  the other codec parts is on a private IP address.

  while I can't call the other party, the other party may call for me, and I wonder how it worked, taking into account the fact that there is no gateway service aware H.323 in the call, either a VCS or aware firewall H.323.

  Experience, firewalls and other gateways outside of Cisco, Tandberg, Polycom, have struggled to deal with the new H.323 version, again this is why I'm puzzled as to why the call worked.

  I did a bit of reading on the new version of H.323 and noticed the option multiplex logical channel, however on a call where I saw this apparently works again of a life-size codec for a Codian MCU 4505 shows no sign of this logical channel multiplex, unless that is named differently in the newspapers that the ITU document calls the function.

  greatly appreciated all all all the answers, I don't understand exactly how the firewall impact VC calls.

  Thinking with portals

  The MXP has NAT builtin functions. Please take a look at the guide admin 9.x:

  http://www.Cisco.com/en/us/docs/Telepresence/endpoint/MXP-series/F9/administration_guide/mxp_series_administrator_guide_f90_excl-full-menu-structure.PDF

  The description of the NAT setting is on page 77.

  EX series admin guide http://www.cisco.com/en/US/docs/telepresence/endpoint/ex-series/tc6/administration_guide/ex-series-administrator-guide-tc62.pdf has the same details on page 63.

• VCS Configuration Start

  Hello

  I'm starting a VCS application to our office Installer.

  Got VCS X7.1 MCU 4505 4.3 (1.68) & CUCM 8.6.2

  First, I had 2 x MX300 and 1 unit of x MX200.

  on th eend CUCM points work very well as 2way. How to check that I have enough licenses for VCS and MCU start configuration. as the insight VCS gives the following result. Like all the configuration is on LAN/WAN not involved any firewall should I these licenses?

  Call non-traversal licenses	Current	0
  	PIC	0
  	Since the last restart	0
  	License limit	0

  ? mediaRouted = true "> call traveled licenses"Current 0 PIC 0 Since the last restart 0 License limit 0 Licenses for registration Current 1 PIC 1 Since the last restart 2 License limit 2500

  The VCS - C should come with 10 non-traversal and 100 licenses of course, unless things have changed since we bought our...

  You will need both; non-traversal for h323<-->h.323 or sip<--->sip calls within the Organization and interworking sip and IPv4<--->IPv6<--->h323 crossing licenses called the organization.

  /Jens

• Questions about VCS, MCU, ISDN gateway license

  Dear Expert.

  I question of license on my next device.

  (1) I have:

  VCSC - 10 calls Non-traversal, crossing 100 calls license license

  vcse - 5 traversal calls

  If I register the mcu and the endpoint to the VCSC as the guardian h323.

  I should have a 15 (5 calls traveral and 10 calls non-traversal) sites h323 conference call simultaneously on the MCU?

  > If I record my MCUS and endpoints to the ISDN gateway as the guardian h323.

  > I should be able to do an 18 sites (1 h323 calls and ISDN 17 * 128 Kbps) conference at the same time call

  > on the MCU?

  GK embarked on telepresence ISDN Gateway has no limitation for the number of concurrent calls.

  Limitation is only for the registration number.

  As gateway ISDN support audio connection gateway exclusively, with simultaneous simple T1 (T1 = 24 b, PRI = 23 b = D), maximum number of calls is superior to '18' as audio call consume only 1 b-channel.

• Question MCU

  Hi guys, I need your help, I made a big mistake, I reset of MCU and after the reboot of the MCU, I lost the product key or feature

  I need to know how can I get a new

  I don't have a contract with Cisco

  I bought the MCU on Ebay!

  I've corrected the link in my previous answer, for some reason when it copied, left part of the URL.

  I have not a way to take a screenshot at the moment, but when you go to this site, look for the option "Obtain other licenses" and in this drop-down list, you must select "License TelePresence to return", then enter the serial number of your MCU.