Telepresence MultiPoint switch

Similar Questions

• Conversationalist active Server of telepresence and Silhouette film strip

  Here's the scenario:

  Three IX5000 usings call conference bridge.  A bunch of users are also challenging a single Meetingplace audio conference.  Then a user IX5000 will merge the telepresence as well as audio Conference and all is well.

  It work very well using CTM (Cisco Telepresence Multipoint switch).

  We recently tried to replace CTMS with telepresence server.  The problem is the silhouette of the active talker band and film are rampant.

  With the server TP when the active talker calls a Phone 7960 series her generic figure filling an IX 5000 screen so it is useless and annoying.

  Also, we get a lot of generics of the silhouette for all callers call professionals is not useful and annoying.

  Can we disable the active speaker and functions the film for points not telepresence endpoints or something?  Or is there another approach to take?

  Hi Tod,

  Unfortunately there is no current way to disable avatar audio active speaker on version 4.1 of the GST and especially beyond manually enter a control center conference TMS and video for the user who is audio only of inhibition.

  What follows is a good thread on this same issue:

  https://supportforums.Cisco.com/discussion/12631876/disable-Telepresence...

  CSCus88355 is used in a feature request to allow this should be disabled, I would advise this sunrise with your account team directly.

  -Jonathan

• TP, video call Cube for the Cube

  Hello world

  I need some advice,

  I have two side;

  First; CCM 6.1.5 + CTMS (Cisco telepresence Multipoint Switch) + Telepresence Cisco 3845 (c3845-adventerprisek9_ivs - mz.124 - 22.YB8.bin) of 1000 +.

  Second; CTMS + TP + CCM 8.5 1000 + 3945 (3900-universalk9-mz. Spa.150 - 1.M4.bin).

  I tried video call on the cube for the cube. I did some configs but will try tomorrow J.

  I need some answers:

  • 3845 router has not mpov, mpov it mandatory for cube calls?
  • 3845 router is it need license for cube?

  Here's the configs;

  3945 router config-

  voice, send rtp-received

  !

  voip phone service

  allow sip to sip connections

  Fax protocol t38 ls-redundancy hs 0-redundancy help 0 no

  SIP

  binding control source-interface GigabitEthernet0/0.21

  bind media source-interface GigabitEthernet0/0.21

  offer-early forced

  !

  voice class codec 1

  Aacld of preference profile 1 Codec 1

  Profile of h264 codec video 2

  !

  !

  interface GigabitEthernet0/0.21

  encapsulation dot1Q 21

  address IP CUBE3945 I_WANIP

  !

  !

  IP forward-Protocol ND

  !

  IP http server

  no ip http secure server

  !

  !

  !

  !

  aacld profile 1 codec

  fmtp "fmtp:96 profile-level-id = 16; streamtype = 5; mode = AAChbr; config = B98C00; sizeLength = 13; indexLength = 3; indexDeltaLength = 3; constantDuration = 480.

  !

  !

  Codec h264 profile 2

  fmtp "fmtp:112 profile-level-id = 4 d 0028; apors-parametersets = R00AKAmWUgDwBDyA, SGE7jyA =; radio-mode = 1.

  !

  !

  Dial-peer voice 1 voip

  Description to_CCM85

  service logon

  RTP payload type cisco-codec-fax-ind 110

  RTP payload type cisco-codec-aacld 96

  RTP payload type cisco-codec-video-h264 112

  session protocol sipv2

  session target ipv4: CCM85_IP

  incoming called number 10...

  codec voice-class 1

  DTMF-relay rtp - nte

  !

  Dial-peer voice 2 voip

  Description to_CCM615

  service logon

  destination-model 7...

  RTP payload type cisco-codec-fax-ind 110

  RTP payload type cisco-codec-aacld 96

  RTP payload type cisco-codec-video-h264 112

  session protocol sipv2

  session target ipv4: 3845cube_wanIp

  codec voice-class 1

  DTMF-relay rtp - nte

  !

  3845 router config-

  voip phone service

  media stream-autour

  allow sip to sip connections

  SIP

  binding control source-interface GigabitEthernet0/1.21

  bind media source-interface GigabitEthernet0/1.21

  offer-early forced

  !

  !

  !

  voice class codec 1

  Aacld of preference profile 1 Codec 1

  Profile of h264 codec video 2

  !

  !

  !

  interface GigabitEthernet0/1.21

  encapsulation dot1Q 21

  address IP CM6_wanIp

  !

  aacld profile 1 codec

  fmtp "fmtp:96 profile-level-id = 16; streamtype = 5; mode = AAChbr; config = B98C00; sizeLength = 13; indexLength = 3; indexDeltaLength = 3; constantDuration = 480.

  !

  Codec h264 profile 2

  fmtp "fmtp:112 profile-level-id = 4 d 0028; apors-parametersets = R00AKAmWUgDwBDyA, SGE7jyA =; radio-mode = 1.

  !

  !

  !

  Dial-peer voice 1 voip

  Description to_CM8

  service logon

  destination-model 10...

  RTP payload type cisco-codec-fax-ind 110

  RTP payload type cisco-codec-aacld 96

  RTP payload type cisco-codec-video-h264 112

  codec voice-class 1

  session protocol sipv2

  session target ipv4:3945_wanIp

  DTMF-relay rtp - nte

  !

  Dial-peer voice 2 voip

  Description to_CM6

  service logon

  RTP payload type cisco-codec-fax-ind 110

  RTP payload type cisco-codec-aacld 96

  RTP payload type cisco-codec-video-h264 112

  codec voice-class 1

  session protocol sipv2

  session target ipv4:CM6_IP

  incoming called-number 7...

  DTMF-relay rtp - nte

  !

  Best, thank you...

  Barris salvation,

  Try adding the following configuration:

  voip phone service
  RTP ssrc multiplex
  address - hide
  allow sip to sip connections
  SIP
  midcall-signalling passthru
  tcp transport session
  passage of Header error-passthru
  amp content sdp
  rel1xx disable

  Thank you

  Paul

• Telepresence - study map Exchange

  Hi all

  I'm going to be assigned to a project of telepresence telepresence telepresence Exchange Server, Multipoint switch and of course Manager using a few video the end points, but no information on who do not yet have. The overall design is that all the components of the solution will be hosted on a private cloud.

  I was wondering if someone could give me, from your experience, the best study roadmap o path of all components that would allow me to have a good knowledge about this technology.

  Thanks in advance for your suggestions.

  Remi

  HII,

  My name is André telepresence support. I would say that whenever you start to study on any technology, you should start from the base first. I think that the best way to do this would be as well to move its base.

  Start of study endpoints and protocols that actually do videoconference take place. Once you get the concept as to how work endpoints, it would be easy for you to climb the ladder and study on the Manager of telepresence and the Multipoint switch.

  So first identify the endpoint with which you work. Once you know this, get the administrative guides for these endpoints and to understand how they work. Then, you can study the Multipoint switch, etc.

  Kind regards

  Saint André

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• TP Installation

  Hello

  I am new to telepresence, and I got a job to establish the environment of Cisco WebEx One Touch Meetinig Teleprence. have got facilities such as below:

  MCS-7835-I3-ECS1, CTS 3000 codec; the documentation seems there is Server/service as Community trade marks, CT-manager and CUCM.  My question is: what equipment should perform such services, or need to connect remote CUCM?

  It would be better if can get more information

  I would recommend starting with the 'prerequisites' here:

  http://www.Cisco.com/en/us/docs/Telepresence/cts_admin/webex_solutions/guide/cts_webex_pref.html#wp1010482

  The WebEx OneTouch feature requires a CTMS (MultiPoint Switch), CTS-Manager & CUCM to implement. You can use the Commerical Express option to combine on a single MCS Server (there are some drawbacks with the number of users, etc.), see here for more details on this:

  http://www.Cisco.com/en/us/partner/docs/Telepresence/cts_admin/Commercial_Infrastructure_Solution/guide/bundle_solutions_info.html

• Call dropped upon joining a scheduled meeting with the bridge

  I am trying to upgrade our infrastructure for telepresence. We want to reassociate CTS-3000 endpoint at all locations new TP servers on newer versions (as compatibility matrix confirms versions are compatible), before you update the firmware of CTS-3000.

  Current firmware of CTS-3000: 1.7.0.2 (4719)

  New servers

  • CUCM 8.6.2a
  • CTS-Manager: 1.9.0.0 (198)
  • CTS Multipoint Switch: 1.9.3.0 (89)

  When the placement of the CTS-3000 on the new server, they register correctly with the 7970 phones that control the CTS.

  Everything works except the most important: at the request of the meetings by using the bridge of Community trade marks.

  • Meeting ad - hoc after the CTMS works,
  • Connection to a static meeting on all CTS MARKS works,
  • When you start a meeting, the first room to compose the bridge will connect ang get the message saying «you are the first participant...» ». When other parts attempt to reach the bridge, the call is dropped. I found in some newspapers (I don't remember those who and cannot find it again) and the error message saying something that he loses the appeal because the line is busy...

  The other thing is that on the new Director of the CTS, when I go to support->-> summary endpoints, in the DUDM column-> device error, there is an 'X' indicating an error. But there is no link with a meesage to error and I don't know where to look. I think this is weired because the CTS-3000 and 7970 recording properly on the CUCM phone.

  I have spend hours to compare the configuration of all servers and the web search, but may not know what is worng. Someone at - it a matter of experience?

  Hi Matt,

  Try this.

  -Connect with CUCM

  -Go to the user of the Application and select the user from CTS - Manager

  -Remove the point of termination/private room with 'X' mark on the subject.

  -Save it and re - discover the room

  -It must be passed to CTS-Manager

  -Now follow the same procedure and add the MAC address of controlled device (section of the users of the App)

  -Rediscover the piece, and then try again.

  Kind regards

  -Sagar

• CTS 3000 on the internet using the VCSC e VCSEX

  Hi all

  One of our clients currently use:

  2 CTS 3000

  1 CTS 1300

  CUCM

  CTS Manager

  Multipoint switch

  He must now make calls through the Internet.

  My questions are:

  Can I use a firewall traversal with VCSC solution in the trunk with the CUCM?

  If I use this solution one CTS can relate to each other on the Internet?

  If he can do this call, the screen is the classic with three monitors?

  Do I have to use the telepresence server to generate the call?

  Thanks in advance

  Best regards

  Sandra

  Hi Sandra,

  You can use the following configuration

  CTS > CUCM > Sip Trunk > VCS-control > VCS-Highway > Internet > endpoint on the Internet.

  Please note that if you use no endpoint 3 screen, the display will be one screen only, because this wont be a negotiated TIP call.

  For the best experience, you can use the telepresence server to allow the appeal and make your CTS and interest user join the GST. So multi-screen experience is preserved.

  Kind regards

  Sagar

• It is possible to intregate MARKS and server of telepresence in a multipoint network link?

  Hi friends,

  I recently did a big upgrade of an environment of telepresence entire Cisco, including a large number of components telerepsence, these VCS, server TP, TMS, CUCM, CTS 1300, CTMS, CUVC, point final series C, jabber, 9900 and 8900 video phones in the series.

  Opportunities for interoperability between Cisco telepresence solutions have been much improved, that's why we have done the upgrade.

  Now, I just doubt, which I wasn't able to solve by searching in the documents and guides. I know that it is possible to establish a link of multipoint network between several CTMSs in order to scale the number of participants at the Conference. It's great! But is it possible to link (waterfall) CTMS (version 1.9) and TP Server (version 2.2)?

  Two of them having supported the ADVANCED Protocol, I think it is supossed to work, right? I tried to achieve this integration, but it did not work. TP recongnizes CTMS as being three SIP screen Telepresence server, I have a participant (CTS Endpoint 1.8) connected to the CTMS and a participant (point endpoint 9971) connected to the server TP, but I see a black screen in both participants, no audio. If endpoints ask each other directly, it works normally, because they are running new firmwares.

  It is possible to intregate MARKS and server of telepresence in a multipoint network connection?

  Thank you!

  Kind regards

  Paulo Souza

  Hello Paulo

  Glad to hear that you are experiencing improved after the upgrade to get the best of interop.

  Unfortunately cascading is not supported with GST again. It can be included in a later version in 2013.

  Thank you

  Rafal

• Firefox does not work for multiple instances of the SAME user account on Windows Multipoint Server 2011

  We have a PC HP of MulitSeat MS6200
  It runs Microsoft Windows MultiPoint Server 2011 (which seems to be a twisted version of Windows 7)
  It is implemented in a laboratory of computer science and students connect you using their account for shared room - IE multiple instances of the same user accounts are currently running on the PC at the same time.
  The first student to run Firefox can work with it without a problem.
  However, when another student try to start firefox they get the following message:
  Firefox is already running but is not responding. To open a new window, you must first close the existing Firefox process, or restart your system.

  Cannot start Firefox using a profile that is already used by someone else.
  Each Firefox instance needs its own profile or you will get this error message.

  • http://KB.mozillazine.org/Using_multiple_profiles_-_Firefox
  • http://KB.mozillazine.org/Bypassing_the_Profile_Manager

  Use-no.-line switch remote control to open another instance of Firefox with its own profile and to different instances of Firefox running concurrently.

  • http://KB.mozillazine.org/Opening_a_new_instance_of_Firefox_with_another_profile

• Creation of models custom TelePresence server on the virtual machine

  Hello

  1. can we develop custom presentations (especially a small picture of a speaker in the corner of a screen presentation of content) in Server TP Cisco on VM?

  2. it is possible to switch between patterns existing and created manually during a live teleconference?

  Thank you, Nick.

  Custom presentations are not supported on server of telepresence, the only currently available layouts are:

  • EQUAL
  • Eminent
  • Overlay
  • Unique

  You can switch between layouts by using DTMF, keys 2 and 8, or ActiveControl if one button 8 or 10 is connected to the endpoint.

• Aironet 1300 - External Point to Multipoint

  Hello world

  I'm looking to expand my network to cover a number of office buildings in an industrial area with a distance of 200 m max to offer 10 MB internet access service supported.

  If I had to mount the Cisco Aironet 1300 on a pole with a visibility of buildings and also provide the same Cisco AP on the side of the customer, from this point to multipoint link (mounted outdoors if possible) would need no additional hardware? Other then DHCP, DNS servers, switch and router.

  Audit miss me not something important!

  Kind regards

  Tom

  I would recommend the 2506. The 24120 would not be a good target for this application. Such a high gain omnidirectional antenna can cause the signal of your remote sites exceeded. And Yes, the internal antennas for remote sites will be just fine. The internal antenna is strong enough and can go at least a kilometre. 200 m is cake for her.

  Best practice is to use the supplied antenna cable length and do not lie down. You can definitely do what you're suggesting, but depending on your length of coaxial cable, you will lose a bit of signal in doing so.

  The double-coaxial (from the bridge to the power injector) and Ethernet cables can each be up to 100 m in length, with no loss. Therefore, recommended would be:

  Antenna mounted to the mast

  Bridge mounted on mast

  <100m of="" double-coax="" (runs="">

  Power Injector (mounted inside)

  <100m of="" ethernet="" (runs="" to="" data="">

  If you want to consider installation of the bridge in your pop and running the antenna cable to the top of the mast, just know what kind of loss, you'll see. Coaxing them manufacturer, Cisco or otherwise, won't say what is the dB/foot or dB/meter, so you can know exactly how much you will lose the length of the cable. I would recommend against the design however, because you might need a larger antenna that will result by distributing a noisy signal.

  Jeff

• TelePresence VLAN

  We have a Manager Communications Group to serve our PBX.  We recently bought a new telepresence system and a 2nd CM cluster that is dedicated to the CTS.  Telepresence SRND States to put the CTS codec in the VLAN voice. There is no mention of this VLAN to the 2nd cluster dedicated to CM in.  I don't think that we can have clusters of 2 CM on the same VLAN voice then what would be the best practice to set the codec and the cluster of CM for the CTS?  Thank you.

  http://www.Cisco.com/en/us/docs/solutions/enterprise/video/tpendpts.html

  Hello

  It's more a question of CUCM SRND, but based on our CUCM SRND take a look at this:

  http://www.Cisco.com/en/us/docs/voice_ip_comm/CUCM/srnd/7x/netstruc.html#wp1043629

  In general, servers of Cisco Unified Communications Manager (Unified CM) clusters, including the multimedia resource servers, reside in a data center or server farm environment. [...] Because these servers and resources are essential to voice networks, Cisco recommends distributing all servers in clusters CM unified, gateways voice hardware and centralized centralized resources among multiple switches physical and, if possible, several physical locations within the campus. This allocation of resources to ensure that, given a hardware failure (for example, a switch or switch line card failure), at least some servers in the cluster will be always available to provide telephony services. In addition, walkways, and material resources will be always available to provide access to the PSTN and to provide auxiliary services.    In addition to being distributed physically, these servers, gateways and material resources should be distributed between the VLANS separated or subnets so that, if a broadcast storm or a denial of service occurs on one VLAN, not all services and voice connectivity will be disrupted.

  ...

  There is the potential for a large number of devices in a single VLAN or broadcast domain to generate large amounts of broadcast traffic periodically, which can be problematic. A good rule of thumb is to limit the number of devices per VLAN to about 512, which equals two subnets of class C (that is, a class C address of subnet 23 bits masked). Typical access layer switches include the stackable Cisco Catalyst 2950, 3500XL, 3550 and 3750, as well as the Cisco 3560 and the larger, high-density Catalyst 4000 and 6000 switches.

  

  Note The recommendation to limit the number of devices in a single VLAN of Communications unified about 512 is not only due to the need to control the amount of broadcast traffic VLAN. For Linux Unified CM-based server platforms, the ARP cache has a limit of 1024 devices. Installation of Unified CM in a VLAN with an IP subnet that contains devices more than 1024 can cause Server Unified CM ARP cache fills up quickly, which can seriously affect the communications between the Unified CM server and other Unified Communications endpoints. Although the size of the ARP cache on server based on Windows Unified CM platforms is developing dynamically, Cisco recommends a limit of 512 devices in any VLAN regardless of the operating system that is used by the Unified CM server platform.

  To answer your question, Yes, separate the VLAN for CUCM, not only for clusters, servers (one cluster contains 1 to 20 servers (super cluster))

  Let us know if you have any questions

• Telepresence C40 intermittent on / off input audio microphone

  Hello

  There was a time that audio input microphone of telepresence C40 goes OFF mode when you start the unit. Also when manually activate us it and end the videoconference call.

  The mode will be automatically extinguished.

  The configuration is C40 entry audio microphone is connected to the mixer of third party.

  Is there no workaround solution to fix?

  Much appreciated for your help. Thank you very much!

  Hi all

  Ralph has opened a case with TAC and arrived at webex. It's just to let everyone know that we were able to solve the problem. Just to convey information. That's what we were doing on the webex session:

  Indeed, we found tha the microphones were shut off as soon as we have disconnected from the call. We checked the following things:

  xConfiguration conference 1 AutoAnswer Mute: Off
  

  xConfiguration MicUnmuteOnDisconnect Mode 1 Conference: on
  

  xConfiguration Provisioning Mode: TMS

  We turned off the MicUnmuteOnDisconnect first and the problem still persists. We have then canceled endpoint of the TMS suspecting that configuration could come from the TMS and checked by calling again. Hoewver, the microphones were always switch off to end the call.

  Finally, we checked the parameters of MuteOnInactiveVideo:

  xConfiguration Audio input Microphone 1 VideoAssociation MuteOnInactiveVideo: on
  

  xConfiguration Audio input Microphone 2 VideoAssociation MuteOnInactiveVideo: on

  We shot both of them off and the problem was solved. Ralph, please check this question as answered so that others can enjoy.

  Kind regards

  Saint André

• Cisco TelePresence SX20 + VCS 8.2 Server and Skype for business 2015

  Nice day! My name is Dimitri! Have Cisco TelePresence SX20 + VCS 8.2 Server and Skype for business 2015. Is it possible to set up a conference call SFB and appeal to all subscribers of this conference? At present, we can only call on a single client and SFB. We have the current version of the software TCNC5.1.4.295090 SX20 - is it possible he obnoviti 7, and it will help to improve the situation? Thank you!

  I'm no expert on this technology, but I have some idea and already deployed some projects dealing with the integration of Cisco (CUCM, VCS) video infrastructure to Microsoft video infra like Lync. It would be the same process to make it work in your case.

  How does your deployed SFB? How do you do a conference for audio and video, participants can join the conference by dialing?

  • Mediation server would be required as a front-end server for you to integrate IP/PBXS using SIP trunk to enable voice (audio) Conference with the participants to the PSTN.
  • AVMCU Should be (Audio video Multipoint Conferencing Unit) will host the video conference that will include video endpoints as participants.
  • Video of Interop (SCREW) Server would be required to interface your SFB to third video conference infrastructure such as Cisco UCM or VCS for the endpoints Cisco call client SFB of endpoint of Cisco to participate to the conference organised on SFB by dialing the conference number.

  I'd say contact your AM SFB or Support as well for queries about how to plan your infrastructure SFB and direction.

  Kind regards

  Acevirgil

• Do I need a CUCM for the implementation of a telepresence system 1300?

  can I install a telepresence system in an office with a UC560 or simply not use any appeals officer in the implementation of a telepresence system 1300?

  Hello

  With CTS endpoints you MUST use cucm to save the endpoints and callmanager will manage the firmware of endpoints, road marking, speed disls, midlet and all the other options and to let the CTS human access to these endpoints for control of booking and call for calls point to point and multipoint

  On the use of VPN recently it is authorized by cisco, according to me, especially for the endpoints one screen you can search for telepresence on wide band for a few cisco whitepapers

  Hope this helps

  If useful rates

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