Cisco Meeting Server 2.1.0 Release Notes...Release Notes,2.1 Release Notes, Cisco Meeting Server...

CiscoMeeting ServerCisco Meeting Server Release 2.1.0Release NotesJanuary 05, 2017

Cisco Systems, Inc. www.cisco.com

http://www.cisco.com/

Cisco Meeting Server Release 2.1.0 : Release Notes 2

Contents

1 Introduction 4

2 New Features/Changes in 2.1 62.1 Call BridgeGroups 6

2.1.1 Load balancing calls acrossCall Bridges 62.1.1.1 How to enable load balancing of calls across a Call BridgeGroup? 72.1.1.2 How to determine themedia loading on a Meeting Server? 72.1.1.3 How to specify the load limits on a cluster of Meeting Servers? 7

2.1.2 Associating Web Bridges, Recorders, Streamers, and TURNservers to CallBridges and Call BridgeGroups 82.1.2.1 How to set which Call BridgeGroups or specific Call Bridgesconnect to the components? 9

2.2 Support for ActiveControl 92.2.1 ActiveControl on theMeeting Server 92.2.2 Limitations 102.2.3 Overview onActiveControl and the iX protocol 102.2.4 DisableUDT within SIP calls 112.2.5 Enabling iX support in Cisco Unified CommunicationsManager 112.2.6 Filtering iX in Cisco VCS 122.2.7 iX troubleshooting 12

2.3 Streaming meetings 132.3.1 Overview of steps to configuring theStreamer 172.3.2 Example of deploying streaming 182.3.3 Streamer licensing 19

2.4 Improvements to the joinmeeting experience for participants using SIP endpoints 202.5 Support for Cisco ExpresswayX8.9 212.6 Miscellaneous changes and improvements 22

2.6.1 Support for multipleCDR receivers 222.6.2 On screenmessaging 222.6.3 Disconnect inactive calls 232.6.4 Improvement to media handling on VMs 232.6.5 Web Bridge 2.0 232.6.6 Incoming calls to Cisco Meeting Apps 232.6.7 Additional voice prompts and background images 242.6.8 Ad Hoc conference license consumption 25

2.7 Summary of MMP changes 25


2.7.1 MMP commands for theStreamer 252.8 Summary of API Additions & Changes 26

2.8.1 Support for grouping Call Bridges 272.8.2 Support for load balancing acrossCall Bridges 272.8.3 Support for streaming meetings 272.8.4 Support for ActiveControl 282.8.5 Support for on screen text 282.8.6 Disable incoming calls to users of Cisco Meeting App 282.8.7 Selecting the joinmeeting experience for participants using SIP endpoints 282.8.8 Other minor additions 28

2.9 Summary of CDRAdditions & Changes 292.10 Cisco endpoints no longer supported 29

3 Notes on Installing and Upgrading to Cisco Meeting Server 2.1 303.1 Upgrading to Release 2.1 303.2 Cisco Meeting Server 2.1 Deployments 32

3.2.1 Deployments using a single host server 323.2.2 Deployments using a single split server hosted on a Core server and an Edge

server 323.2.3 Deployments for scalability and resilience 33

3.3 Downgrading 33

4 Resolved Issues 34Resolved inMeeting Server 2.1.0 34

5 Known Limitations 35

Cisco Legal Information 38

Cisco Trademark 39


1 IntroductionThis release note describes the new features, improvements and changes in release 2.1.0 of theCisco Meeting Server software for Cisco Meeting Server 1000, Acano X-Series Servers, andvirtualized deployments.

TheCisco Meeting Server was formerly called theAcano Server. TheCisco Meeting Server canbe hosted on:

n theCisco Meeting Server 1000, a Cisco UCS server preconfigured with VMware and theCisco Meeting Server installed as a VM deployment.

n theAcano X-Series hardware,

n or on a specification based VM server.

TheCisco Meeting Server software is referred to as theMeeting Server throughout theremainder of this guide.

If you are upgrading from 2.0.x, you are advised to take a configuration backup using thebackup snapshot <filename> command, and save the backup safely on a differentdevice. See theMMPCommand Reference document for full details.

Note about XMPP resiliency: the XMPP resiliency feature first introduced in Acano Server release1.8 is a fully released feature in Cisco Meeting Server 2.1, and supported for productiondeployments.

Note about SIP edge: TheCisco ExpresswayX8.9 supports traversal of SIP traffic at the edge ofthe network, to and from theCisco Meeting Server.

You are encouraged to evaluate theSIP edge support provided by the Expressway. In a futurerelease theSIP edge feature in theMeeting Server will bewithdrawn. For more information, seeSection 2.5.

TheSIP and Lync Call Traversal feature first introduced in Acano Server release 1.8, is still a betafeature in Cisco Meeting Server 2.1.0, and is not intended for a production environment.

Cisco does not guarantee that a beta featurewill becomea fully supported feature in the future.Beta features are subject to changebased on feedback, and functionalitymay change or beremoved in the future

Note about rebranding the background image to the login page for theWebRTC app:MeetingServer 2.1 supports the redesignedWeb Bridge 2.0.Web Bridge 2.0 does not supportrebranding the background image for the login page to theWebRTC app. For more information,seeSection 2.6.5.

1 Introduction


Note about messageboard chat: For existing deployments that usemessageboard chat, chatwill remain enabled when you upgrade to 2.1. Otherwise, youwill need to use theAPI to create acallProfilewith parameter messageBoardEnabled set to true.

1 Introduction


2 New Features/Changes in 2.1Release 2.1 of theMeeting Server software comprises:

n Call BridgeGroups and load balancing calls

n ActiveControl

n streaming meetings

n improved join options for meetings

n support for Cisco ExpresswayX8.9

n a few miscellaneous new features

n additional MMP commands

n additional API objects and parameters to support these new features

n additional CDR support for new features.

n Cisco endpoints no longer supported

Note: The term spaces is used throughout the documentation apart from theAPI guidewhichstill uses the old terminology of coSpaces.

2.1 Call Bridge GroupsA typical large scale deployment consists of several Meeting Servers located at multipleoffices/data centres. To minimise network load, reduce firewall configuration and to ensureefficient use of theCall Bridge resources, it is now possible to configure location information forcomponents. A location could refer to a single datacentre, or a continent. The decision of howto group Call Bridgeswill depend on the specifics of your network configuration and the desiredbehavior.

Version 2.1 of theCisco Meeting Server software introduces theAPI object/callBridgeGroups to specify a group. It also introducesAPI fields to limit the usage of othercomponents to either a specific Call Bridge or Call BridgeGroup.

2.1.1 Load balancing calls acrossCall Bridges

Ideally all of themedia for calls to a conference should reside on the sameCall Bridge if users arein the same location and if the required call capacity exists.When users are inmultiple locationsthen ideally oneCall Bridge per location should be used.

Creating a Call BridgeGroup with Call Bridges that are configured as a cluster, will result inintelligent load balancing of calls across theCall Bridges in the cluster. For the load balancingfeature to work correctly, a Round Trip Time (RTT) of less than 100ms is required for the servers

2 New Features/Changes in 2.1


in a Call BridgeGroup. Themaximum RTT between any two nodes in the samecluster remainsas 300ms.

If theCall Bridges in a group are heavily loaded, then calls can bemoved to Call Bridges in adifferent group using a call control device such as theCisco Unified CommunicationsManager.The intelligent decisionmaking behind where calls end up, is handled by theMeeting Servers.The call control system needs to be able to handle SIPmessages from theMeeting Servers andmove calls to the correct location. This functionality has been tested using Cisco UnifiedCommunicationsManager, which is the only Cisco supported call control system for thisfunctionality. TheCisco VCS is not currently supported since it doesn’t include support forINVITEwith Replaces.

By default, a Call Bridge in a Call BridgeGroup will reject all calls from new participants at 80%load, and only new distribution callswill be allowed. Thewhite paper entitled “LoadingBalancing AcrossCisco Meeting Servers” explains how load balancing is implemented acrossCall Bridgeswhich are in a Call BridgeGroup. It provides examples of how Call BridgeGroupscan be used to redirect calls if particular Call Bridges are heavily loaded. It also explainswhat isrequired in a dial plan to implement call redirection.

2.1.1.1 How to enable load balancing of calls across a Call BridgeGroup?

Perform a PUT on the new API object /callBridgeGroupswith theloadBalancingEnabled parameter set to true.

2.1.1.2 How to determine themedia loading on aMeeting Server?

Perform a GET on the new API object /system/load. A numeric value for parametermediaProcessingLoadwill be returned, this represents the load on theMeeting Server

If you haveCall BridgeGroups configured, and you have load balancing activated, then callsfrom new participants are rejected at 80% load.

If you are not using load balancing with Call BridgeGroups, then callswill not be rejected, butthe quality of all callswill be reduced when the load limit is reached. If this happens often, werecommend that you buy additional hardware.

Tip: If you have only oneCall Bridge, and youwant to reject calls rather than reducing quality,you can create a Call BridgeGroup with a singleCall Bridge and enable load balancing.

2.1.1.3 How to specify the load limits on a cluster of Meeting Servers?

Perform a PUT on theAPI object /system/configuration/clusterwith the followingparameters set:

n loadLimitwith a numeric value for themaximum load on theMeeting Server

Suggested Load limits.



System Load Limit

CMS1000 96000

X3 250000

X2 125000

X1 25000

VM 1250 per vCPU

Note: These load limits are currently being evaluated and may change.

2.1.2 AssociatingWeb Bridges, Recorders, Streamers, and TURNservers toCallBridges and Call BridgeGroups

From version 2.1,Web Bridges, Recorders, Streamers, and TURNservers can be associatedwith individual Call Bridges and Call BridgeGroups. If a component is configured with either aCall BridgeGroup or a Call Bridge, then only theCall Bridges in the group or the specific CallBridgewill attempt to connect to the component.

For instance:

n aWeb Bridgewith a callBridgeGroup set: only theCall Bridges in theCall BridgeGroup thatwas set for theWeb Bridge, will attempt to connect to theWeb Bridge.

n aWeb Bridgewith a specific callBridge set (but no group): only theCall Bridge that was setfor theWeb Bridge, will attempt to connect to theWeb Bridge.

n aWeb Bridgewith neither a callBridgeGroup nor a callBridge set: anyCall Bridgemay attemptto connect to theWeb Bridge.



Figure 1: Associating Web Bridges with Call Bridges and Call Bridge Groups

In Figure 1 above:

n Call Bridge 1 and Call Bridge 2 form Call BridgeGroup 1, andWeb Bridge 1 andWeb Bridge2 are associated with Call BridgeGroup 1.

n Web Bridge 3 hasCall Bridge 3 set.

n Web Bridge 4 has no Call BridgeGroup or Call Bridge set, and therefore anyCall Bridge (CallBridge 1, Call Bridge 2, Call Bridge 3 or Call Bridge 4) may attempt to connect to Web Bridge4.

2.1.2.1 How to set which Call BridgeGroups or specific Call Bridges connect to thecomponents?

Perform a PUT on theAPI objects/webBridges, /recorders, and /turnServerswith thefollowing parameters set: ID of thecallBridgeGroup and callBridge associated with thecomponent.

2.2 Support for ActiveControlFrom version 2.1, theMeeting Server supports ActiveControl for hosted calls. For participantsusing a Cisco SX,MX or DX endpoint with CE 8.3+ software installed, ActiveControl allows themeeting participant to receive details of themeeting and perform a few administrative tasksduring themeeting, using the endpoint interface.

2.2.1 ActiveControl on theMeeting Server

TheMeeting Server supports sending the following meeting information to ActiveControlenabled endpoints:



n Participant list (also known as the roster list) so that you can see the names of the otherpeople in the call and the total number of participants,

n indicator of audio activity for the currently speaking participant,

n indicator of which participant is currently presenting,

n Indicators telling whether themeeting is being recorded or streamed, and if there are anynon-secure endpoints in the call,

n on screenmessagewhichwill be displayed to all participants, seeSection 2.6.2,

In addition, theMeeting Server can control the following features on ActiveControl enabledendpoints:

n select the layout to be used for the endpoint,

n disconnect other participants in themeeting, seeSection 2.8.4

Note: These features are configured using theAPI of theMeeting Server, see defaultLayoutparameter on theAPI objects: /calls, /callLegProfile and /coSpace.

2.2.2 Limitations

n If an ActiveControl enabled call traverses a Unified CM trunkwith a Unified CM version lowerthan 9.1(2), the call may fail. ActiveControl should not be enabled on older Unified CM trunks(Unified CM 8.x or earlier).

n ActiveControl is a SIP only feature. H.323 interworking scenarios are not supported.

Note: ActiveControl usesUDT transport for certain features, for example sending roster lists toendpoints and allowing users to disconnect other participantswhile in a call. SeeSection 2.2.4for the steps to follow on theMeeting Server.

2.2.3 Overview on ActiveControl and the iX protocol

ActiveControl uses the iX protocol, which is advertised as an application line in theSIP SessionDescription Protocol (SDP). TheMeeting Server automatically supports ActiveControl, and thefeature cannot be disabled. In situationswhere the far end network is not known or is known tohave devices that do not support iX, it may be safest to disable iX on SIP trunks between theMeeting Server and the other call control or Video Conferencing devices. For instance:

n for connections to Unified CM 8.x or earlier systems the older Unified CM systemswill rejectcalls from ActiveControl-enabled devices. To avoid these calls failing, leave iX disabled onany trunk towards theUnified CM 8.x device in the network. In caseswhere the 8.x device isreached via a SIP proxy, ensure that iX is disabled on the trunk towards that proxy.



n for connections to third-party networks. In these cases there is no way to know how thethird-party networkwill handle calls from ActiveControl-enabled devices, the handlingmechanismmay reject them. To avoid such calls failing, leave iX disabled on all trunks tothird-party networks.

n for Cisco VCS-centric deploymentswhich connect to external networks or connect internallyto older Unified CM versions. From Cisco VCS X8.1, you can turn on a zone filter to disable iXfor INVITE requests sent to external networks or older Unified CM systems. (By default, thefilter is off.)

2.2.4 DisableUDTwithin SIP calls

ActiveControl uses theUDT transport protocol for certain features, for example sending rosterlists to endpoints, allowing users to disconnect other participantswhile in a call, and inter-deployment participation lists. UDT is enabled by default. You can disableUDT for diagnosticpurposes, for example if your call control does not useUDT, and you believe this is the reasonthe call control does not receive calls from theMeeting Server.

Using theMeeting Server API:

1. Create a compatibility profilewith the sipUdt parameter set to “false”. Either POSTsipUdt=false to the/compatibilityProfiles object or PUT to/compatibilityProfiles/<compatibility profile id> object

2. Disable the use of UDT at the system level, by adding the compatibilityProfile parameterand id (from step 1) to the system profile. PUT compatibityProfile=<compatibility profileid> to the/system/profiles/ object.

2.2.5 Enabling iX support in CiscoUnified CommunicationsManager

Support for the iX protocol is disabled by default in Cisco Unified CommunicationsManager. Toenable iX support, youmust first configure support in theSIP profile and then apply that SIPprofile to theSIP trunk.

Configuring iX support in a SIP profile

1. ChooseDevice > DeviceSettings > SIP Profile. The Find and List SIP Profileswindowdisplays.

2. Do one of the following:

a. To add a new SIP profile, click Add New.

b. To modify an existing SIP profile, enter the search criteria and click Find. Click thenameof theSIP profile that youwant to update.

TheSIP Profile Configurationwindow displays.

3. Check the box for Allow iX ApplicationMedia



4. Make any additional configuration changes.

5. Click Save

Applying theSIP profile to a SIP trunk

1. ChooseDevice > Trunk.

The Find and List Trunkswindow displays.

2. Do one of the following:

a. To add a new trunk, click Add New.

b. To modify a trunk, enter the search criteria and click Find. Click the nameof the trunkthat youwant to update.

The TrunkConfigurationwindow displays.

3. From theSIP Profile drop-down list, choose the appropriate SIP profile.

4. Click Save.

5. To update an existing trunk, click ApplyConfig to apply the new settings.

2.2.6 Filtering iX in CiscoVCS

To configure theCisco VCS to filter out the iX application line for a neighbor zone that does notsupport the protocol, the zonemust be configured with a custom zoneprofile that has theSIPUDP/IX filter mode advanced configuration option set to On.

To update advanced zone profile option settings:

1. Create a new neighbor zone or select an existing zone (Configuration > Zones > Zones).

2. In theAdvanced parameters section, for Zone profile, chooseCustom if it is not alreadyselected. The zone profile advanced configuration options display.

3. From theSIP UDP/IX filter modedrop-down list, chooseOn.

4. Click Save.

2.2.7 iX troubleshooting

Table 1: Call handling summary for calls that contain an iX header

Scenario Outcome

Unified CM 8.x or earlier Calls fail

Unified CM 9.x earlier than 9.1(2) Calls handled normally but no ActiveControl

Unified CM 9.1(2) Calls handled normally plus ActiveControl

Endpoint - no support for iX and no SDP implementation Endpoint may reboot or calls may fail



2.3 StreamingmeetingsVersion 2.1 adds a new component: Streamer, to theMeeting Server. TheStreamercomponent adds the capability of streaming meetings held in a space to theURI configured onthe space.

An external streaming server needs to be configured to be listening on this URI. The externalstreaming server can then offer live streaming to users, or it can record the live stream for laterplayback.

Note: Several standards based streaming servers are known to workwith theStreamer, butCisco only offers support for VBrick as external streaming server.

TheStreamer connects to an external server using RTMPwith an overall bitrate of 2Mbps. Thevideo is encoded using H.264 at 720p30, while the audio is 64kbpsAAC-LC. All traffic betweentheStreamer and the external streaming server is unencrypted.

TheStreamer should be hosted on another Meeting Server instance than the server hosting theCall Bridge, see Figure 2. If the Streamer is hosted on the same server as theCall Bridge (local),then it should only be used for testing purposes or for very small deployments.

The recommended deployment for production usage of theStreamer is to run it on a separateVM. This VM should be sized with 1 vCPU and 1GBof memory per 6 concurrent streams, with aminimum of 4vCPUs and amaximum of 32vCPUs.

Note: TheseVM specifications are currently being evaluated, and the sizes are likely to bereduced.

For more details on VM specification seeUnified Communications in a Virtualized Environment –Cisco (www.cisco.com/go/uc-virtualized).

Where possible, it is recommended that theStreamer is deployed in the samephysical localityas theCall Bridge to ensure low latency and high network bandwidth. If there are networkconnection issues between theCall Bridge and theStreamer, then the resultant stream could beaffected.

Note: youmay need to open firewall ports if the streaming destination URIs are on the externalside of a firewall.


http://docwiki.cisco.com/wiki/Unified_Communications_in_a_Virtualized_Environment


Figure 2: Permitted deployment for streaming: remote mode

TheStreamer also supports redundant configurations, see Figure 3, Figure 4, Figure 5 andFigure 6. If you usemultiple streamers then the solution load balances between availablestreaming devices. To restrict the use of specific Streamers to specific Call Bridges use theCallBridgeGroup functionality introduced in version 2.1.



Figure 3: Permitted deployments for streaming: multiple streamers

Figure 4: Permitted deployments for streaming: Call Bridge cluster

If your deployment hasmultipleCall Bridges and multiple Streamers then everyCall Bridgewilluse every Streamer (see Figure 5), unless the callBridgeGroup and callBridge parameters have



been set for each Streamer using theAPI to PUT to /streamers/<streamer id> (see Figure6).

Figure 5: Permitted deployments for streaming: Call Bridge cluster withmultiple Streamers and no CallBridge Groups set up



Figure 6: Permitted deployments for streaming: Call Bridge cluster withmultiple Streamers and a CallBridge Group and Call Bridge set up

For testing purposes or for small deploymentswith only occasional usage of theStreamer, theStreamer can be co-located on the same server as theCall Bridge. Thismay support between 1to 2 simultaneous streamings.

Note: Acano X series servers used in the single combined deployment mode should only beused for testing theStreamer, they should not be used in production networks to host theStreamer.

2.3.1 Overview of steps to configuring theStreamer

n UseMMP commands to configure and enable theStreamer on a Meeting Server and to addcertificates.

n Use theAPI of theMeeting Server hosting theCall Bridge to configure the settings throughwhich theCall Bridgewill communicatewith theStreamer, and where to save thestreamings.

n Use the new streamingMode parameter on theAPI object /callProfiles or/callProfiles/<call profile id> to select whether a meeting can be streamed ornot.



2.3.2 Example of deploying streaming

Note: TheStreamer behaves as an XMPP client, so the XMPP server needs to be enabled on theMeeting Server hosting theCall Bridge.

This example gives the steps to deploy a streamer remote to theCall Bridge. It assumes that youalready have a working Call Bridge and XMPP server.

1. Create a certificate and private key for theStreamer, following the steps described in theCertificates guidelines for an internal CA signed certificate.

2. SSH into theMMPof theMeeting Server hosting theStreamer.

3. Configure theStreamer to listen on the interface(s) of your choicewith the followingcommand:streamer listen <interface[:port] whitelist>

TheStreamer can listen onmultiple interfaces, e.g. one on public IP and one on the internalnetwork. (However, it cannot listen onmore than one port on the same interface.)

The following is an examplewhere interfaces are set to interfaceA and B, both using port8443.streamer listen a:8443 b:8443

To use a local Streamer, the Streamermust listen on the loopback interface lo:8443, forexamplestreamer listen lo:8443 b:8443

4. Upload the certificate file, key file and certificate bundle to theMMP via SFTP.streamer certs <keyfile> <certificatefile> [<crt-bundle>]

5. Add theCall Bridge certificate to theStreamer trust store using the command:streamer trust <crt-bundle>

6. Use the streamer command to list the details for the streamer, for example:cms1> streamerEnabled : trueInterface whitelist : a:8445 b:8445Key file : streamer0.keyCertificate file : streamer0.cerCA Bundle file : streamer.crtTrust bundle : callbridge.crt

7. Enable theStreamer:streamer enable

8. CreateDNS A record for theStreamer and set it to resolve to the IP Address of the Ethernetinterface youwant theStreamer to listen on.



9. Use theAPI of theMeeting Server hosting theCall Bridge to configure the settings throughwhich theCall Bridgewill communicatewith theStreamer.

a. Specify theHTTPS URL address that theCall Bridgewill use to reach this streamer.Either POST theURL to the/streamers object or PUT to the/streamers/<streamer id> object

Note: If using a local Streamer, theURLmust be the loopback interface, for examplehttps://127.0.0.1:8443

b. POST to /coSpaces or PUT to /coSpaces/<coSpace id> thestreamUrlwhichdetermineswhere streaming is streamed to, if streaming is initiated

c. Select whether a meeting can be streamed or not and whether the streaming will startwithout any user intervention. Use thestreamingMode parameter on theAPI object/callProfiles or /callProfiles/<call profile id>

Options for this are:

automatic - streaming occurswithout any user intervention, if streaming cannot occurthemeeting still occurs.

manual - users canmanually start and stop the streaming using DTMF.

disabled - no users can stream.

d. Control which users have permission to start and stop streaming. Use thestreamingControlAllowed parameter on /callLegProfiles

e. For each space that a user would like to stream, POSTor PUT to /coSpaces thestreamURL parameter specifying the destination URL to stream to.

Note: some streaming services require usernameand password, others provide aunique stream key. For example, for vBrick:streamUrl=rtmp://<username>:<password>@<vbrickIP/FQDN>/live/PullStream1

and for YouTube:streamUrl=rtmp://a.rtmp.youtube.com/live2/<stream key>

f. Use thestartStreaming and stopStreaming parameters for /dtmfProfilesand /dtmfProfiles/<dtmf profile id> to map theDTMF tones for starting andstopping streaming. For example: **7 to start and **8 to stop streaming.

2.3.3 Streamer licensing

Youwill need one ormore licenses for streaming which is loaded on theMeeting Server hostingtheCall Bridge, not the server hosting theStreamer. One ‘recording’ license supports 1



concurrent streaming or 1 recording, existing recording licenceswill allow streaming. Fromversion 2.1, a starter kit is availablewhich includes one recording/streaming license or additionalports. Contact your Cisco sales representative or partner to discuss your licensingrequirements.

2.4 Improvements to the joinmeeting experience for participants using SIPendpointsIn releases prior to 2.1, it was possible to havemultiple accessmethods that shared a URI, buteach had to have a unique non-empty PIN. In 2.1, it is possible to mix PIN and no-PINwith thesameURI. For instance, from version 2.1 it is possible to have separate host and guest PINs, withthe host having a non-empty PIN and guests having an empty PIN. Guests have to press “#”(pound) to join themeeting or, if configured, guests canwait a specified amount of time to jointhemeeting.

To select the option to require guests to press ‘#’ (pound) to join a meeting, set thepasscodeMode parameter to required on /callProfiles/<call Profile id>

To select the option to automatically connect guests after a specified waiting time, set thepasscodeMode parameter to timeout on /callProfiles/<call Profile id> andconfigure the value of the timeout via thepasscodeTimeout parameter on/callProfiles/<call Profile id>.

To support these new combinations of URIs and PINs, and alter the joinmeeting experience forparticipants using SIP endpoints, two additional voice prompts and two additional backgroundimages are available for customization, see Table 2. If these additional files are not included inthe branding archives, then the voice prompts and images used for passcode_entrywill be usedinstead.



Table 2: Join Options for Meetings

Scenario passcodeMode BehaviorBackgroundused

Voiceprompt Notes

All accessmethodshavepasscodes

NA Userprompted forpasscode,must enter ‘#’

passcode_background.jpg

passcode_entry.wav

Used when below filesnot present. If not inarchive then blackbackground and no voiceprompt.

Someaccessmethodshavepasscodes

required Userprompted forpasscode,must enter ‘#’

passcode_or_blank_required_background.jpg

passcode_or_blank_required_entry.wav

If not in customizationarchive then top caseused.

Someaccessmethodshavepasscodes

timeout If user entersnothing, willjoin as ifentered just‘#’

passcode_or_blank_timeout_background.jpg

passcode_or_blank_timeout_entry.wav

If not in customizationarchive then top caseused.

Nopasscode

NA User joinswithout anyadditionalinput

NA NA

2.5 Support for Cisco Expressway X8.9TheCisco ExpresswayX8.9 supports traversal of SIP traffic at the edge of the network, to andfrom theCisco Meeting Server. This allows collaboration using Cisco Meeting spaces betweenon-premiseCisco Meeting App or SIP endpoint users, and users external to the networkwhoare using standards-based SIP endpoints, Microsoft Skype for Business or Microsoft Office365. Cisco Expressway does not currently support traversal for external Cisco Meeting Appusers. Cisco ExpresswayX8.9 is also previewing a Cisco Meeting Server web proxy to enableoff-premise users to joinmeetings held in spaces using a web browser supporting WebRTC.

To use theCisco ExpresswayX8.9 for TURN, rather than the TURNServer in theMeeting Server:

n ignore the TURNconfiguration section in the chapter on Configuring theMMP in theMeetingServer deployment guide. If you have already configured the TURNserver, then disable it viatheMMP command turn disable

then either:

n use theWeb Admin interface of theCisco Meeting Server. Go to Configuration>General andtype the Expressway IP address in the TURNServer address (server) field. TheCisco MeetingServer will use port 3478 to communicatewith theCisco Expressway.



or:

n use theCisco Meeting Server API object /turnServers and set type=expressway.

For more information, see theCisco ExpresswayX8.9 release notes.

2.6 Miscellaneous changes and improvementsRelease 2.1 supports the following changes and new features:

n support for up to four CDR receivers,

n support for on-screenmessaging,

n disconnect inactive calls,

n improvement to media handling on VMs,

n support for Oracle Internet Directory (LDAP version 3)

n support forWeb Bridge 2.0,

n incoming calls to theCisco Meeting App can bedisabled,

n additional voice prompts and background images to allow new combinations of URI’s andpasscodes,

n Ad Hoc conferencesmight consumeonePMP+ license rather than an SMP+ license.

2.6.1 Support for multipleCDR receivers

From version 2.1, theMeeting Server supports up to four CDR receivers, enabling you to deployup to four different management tools, or duplicate instances of the samemanagement tool forresiliency.

Note: The list of CDR receivers is held locally to an individual Call Bridge, it is not stored in thedatabase shared between clustered Call Bridges.

To configure themultipleCDR receivers, POST each URI to theAPI object:

/system/cdrReceivers/<CDR receiver id>

or alternatively, configure themultipleCDR receivers through theWeb Admin Interface, navigateto Configure> CDR settings, enter each receiver’s HTTP or HTTPS URI.

2.6.2 On screenmessaging

From version 2.1, theMeeting Server provides the ability to display an on-screen text messageto all participants in a meeting hosted on theMeeting Server; only onemessage can be shownat a time. The duration that themessage is displayed can be set, or madepermanent until a newmessage is configured.



For users of SIP endpoints and Lync/Skype for Business clients, the on-screen text message isdisplayed in the video pane. The position of themessage in the video pane can be selected fromtop,middle or bottom.

For theWeb Bridge 2.0WebRTC app, themessage is sent out-of-band and the app renders itlocally. On screenmessaging is also sent to other devices that are using ActiveControl in thedeployment, for instanceCE8.3 endpoints, and individual Meeting Servers not in a cluster butwith the in-call message feature enabled.Meeting Servers in a cluster also support on screenmessaging through a proprietarymechanism.

Use themessageText, messagePosition and messageDuration parameters for APIobject /calls. SeeSection 2.8 for more details

2.6.3 Disconnect inactive calls

SIP sessions between theMeeting Server and a call control device, for exampleCisco UnifiedCommunicationsManager, can remain in place even if there is no longer any activity in the call.This situation can arise from a laptop battery dying while in the call, or from network problems.

From version 2.1, theMeeting Server will disconnect and end SIP callswhen theMeeting Serverdetects no media activity in the call over a period of 60 seconds. This includes Lync and Skypefor Business (S4B) callswith no media activity. For SIP, TIP, Lync and S4Bcalls that go on hold ,if the call stops sending RTP/RTCP traffic, then after 60 minutes the call is disconnected, this isto prevent calls hanging around indefinitely.

2.6.4 Improvement tomedia handling on VMs

In version 2.1 of the VM software release, themedia code is isolated from the rest of theCallBridge code. Thismeans that themedia code can be restarted without dropping ongoing callsor loss of any other functionality. Therewill simply be a brief pause inmedia during a restart.

2.6.5 Web Bridge 2.0

From version 2.1, theMeeting Server supportsWeb Bridge 2.0 which has the followingchanges:

l new look and feel for theWeb RTC app.

l the background image for login to theWebRTC app can no longer be customized. Formore information, refer to theCustomization guidelines.

Note about browser to usewithWebRTC App:We strongly recommend only using themostrecent version of Chrome, see this FAQ.

2.6.6 Incoming calls toCiscoMeeting Apps

FromMeeting Server version 2.1, incoming calls to theCisco Meeting App can bedisabled


https://kb.acano.com/content/37/4/en/what-versions-of-browsers-do-we-support-for-webrtc.html?highlight=browser#content


By default incoming calls to Cisco Meeting Apps are allowed, however this behavior can bechanged so that incoming calls are not allowed to users of theCisco Meeting App. Follow thesesteps:

Either:

1. Login to theWeb Admin interface of theMeeting Server, go to Configuration>CMA usersettings.

2. Set Allow incoming calls to not allowed and select Submit.

or use theAPI object to either POST to /userProfiles or PUT to /userProfiles/<user profile id> therequest parameter canReceiveCalls = “false”.

2.6.7 Additional voice prompts and background images

Version 2.1 supports additional voice prompts and background images to allow newcombinations of URI’s and passcodes. These alter the joinmeeting experience for participantsusing SIP endpoints.

The new voice prompts are:



Filename Text of message Repeating? Played when ...

passcode_or_blank_required_entry.wav

Please enter the PIN,followed by the '#'(pound) key.

No a PIN is required for the host to enter the spaceas host, but guests only need to use the #(pound) key.

passcode_or_blank_timeout_entry.wav

Please enter the PIN,followed by the '#'(pound) key.

No a PIN is required for the host to enter the spaceas host, but guests join after a short timeout.

The new background images are:

Filename to use Image used when ....

passcode_or_blank_required_background.jpg

Screen can be shown when a PIN is required for the host to enter the space ashost, but guests only need to use the # (pound) key.

passcode_or_blank_timeout_background.jpg

Screen can be shown when a PIN is required to enter the coSpace as host, butguests join after a short timeout.

New API parameters of passcodeMode and passcodeTimeout for /callProfiles and/callProfiles/<call profile id> are provided to select the joinmeeting experience.SeeSection 2.8.7.

2.6.8 Ad Hoc conference license consumption

Before release 2.1, Ad Hoc conferences never consumed PMP+ licenses.With this 2.1 release,the initiator of theAd Hoc conference can be identified and if they have been assigned a PMP+license then that is used for the conference.

2.7 Summary of MMP changesVersion 2.1 supports these additional MMP commands.

2.7.1 MMPcommands for theStreamer

Command Description

streamer restart Restarts the Streamer

streamer Displays the current configuration of the Streamer

streamer listen <a|b|c|d|lo|none[:<port>] whitelist>streamer listen a b

Sets up the interface(s) and port(s) for the Streamer tolisten on. You must enable the service to start listeningwith the command recorder enable. The default for theoptional port argument is 443.

streamer listen none Stops the Streamer listening.



Command Description

streamer (enable|disable) Enables or disables the Streamer. You need to disable theStreamer before configuring it. After configuration, youneed to enable the Streamer.

streamer certs <keyfile-name> <crtfilename> [<crt-bundle>]

Provides the name of the key file and .crt file for theStreamer and, optionally, a CA certificate bundle asprovided by your CA

streamer certs none Removes certificate configuration

streamer trust <crt-bundle|crt-file> Controls which Call Bridge instances are allowed toconnect to the Streamer.If the trusted Call Bridge is running on the same server asthe Streamer, then issuing the streamer trust commandwith the name of the Call Bridge publiccertificate/certificate bundle is sufficient. If the Call Bridgeis running on another server, the publiccertificate/certificate bundle of the Call Bridge must firstbe copied to the server with the enabled Streamer usingSFTP.

streamer trust none Deconfigures any trust settings

2.8 Summary of API Additions & ChangesNew API functionality for theMeeting Server 2.1 includes support for:

n grouping Call Bridges

n load balancing of calls acrossCall Bridges

n streaming calls

n ActiveControl

n on screenmessaging

n disable incoming calls to users of Cisco Meeting App

n altering the joinmeeting experience for participants using SIP endpoints

there are also someother minor additions.

You are advised not to use beta features in a production environment. Only use them in a testenvironment until they are fully released.



2.8.1 Support for grouping Call Bridges

n New API object to enable the grouping of Call Bridges : /callBridgeGroups

Note: Load balancing calls acrossCall Bridges in a group is disabled by default.

n New request parameter to /callBridges : callBridgeGroup

n New request parameters to /recorders : callBridge, callBridgeGroup

n New request parameters to /turnServers : callBridge, callBridgeGroup

n New request parameters to /webBridges : callBridge, callBridgeGroup

n New failure reason : callBridgeGroupDoesNotExist

2.8.2 Support for load balancing acrossCall Bridges

n New request parameter to /callBridgeGroups: loadBalancingEnabled

n New request parameters to /system/configuration/cluster: loadLimit,newConferenceLoadLimitBasisPoints,existingConferenceLoadLimitBasisPoints

n New API object: /system/load that returns a numeric value for mediaProcessingLoad

2.8.3 Support for streaming meetings

n New API object to enable the streaming of meetings hosted on theMeeting Server :/streamers

n New request parameter to /coSpaces : streamUrl

n New request parameter to /calls : streaming

n New request parameter to /callProfiles : streamingMode

n New request parameter to /callLegProfiles : streamingControlAllowed

n New request parameters to /dtmfProfiles : startStreaming, stopStreaming

n New response value for /calls/<call id> : streaming

n New status value returned on /callLegs/<call leg id> : streaming

n New alarm type for /system/alarms: streamer unavailable

n New response value for features field of /system/licensing : streaming

n New failure reasons : callStreamingCannotBeModified, streamerDoesNotExist,streamingLimitReached



2.8.4 Support for ActiveControl

n New request parameter to /callLegProfiles : disconnectOthersAllowed. POST to/callLegProfiles to create a new call or PUT to /callLegProfiles/<call legprofiles id> if modifying an existing call.

The setting determineswhether participants can drop others from a call when they are usingan endpoint that can support ActiveControl.

The default setting is true.

n New status section returned on /callLegs/<call leg id> : activeControl

Note: If ActiveControl has been negotiated with the remote party, the callLeg informationreturned will include an activeControl section.Within that section, you can seewhethertheActiveControl connection is encrypted.encrypted: true - an encrypted ActiveControl has been negotiated with the remote party.encrypted: false - ActiveControl has been negotiated with the remote party, but it is notencrypted.

n New request parameter to /compatibilityProfiles and/compatibilityProfiles<compatibility profile id>: sipUdt

2.8.5 Support for on screen text

n New request parameters to /calls : messageText, messagePosition,messageDuration.POST to /calls to create a new call or PUT to /calls/<call id> if modifying anexisting call.

n New response value for /calls/<call id> : messageText, messagePosition,messageDuration, messageTimeRemaining

Note: a message can bepermanently displayed.

2.8.6 Disable incoming calls to users of CiscoMeeting App

n New request parameter to /userProfiles/<user profile id>: canReceiveCalls

2.8.7 Selecting the joinmeeting experience for participants using SIP endpoints

New request parameters to /callProfiles and /callProfiles/<call profile id>:passcodeMode, passcodeTimeout

2.8.8 Other minor additions

To support a user (imported through AD) being added as owner of a space:



n added new request parameter to /coSpaces : ownerADGuid

To support retrieval of meeting entry details for a specific space:

n added “meetingEntryDetail” node. Perform a GET on /coSpaces/<coSpaceid>/meeting EntryDetail to obtain the uri and CallId.

To improve filtering on /users:

n added emailFilter, which if supplied, will restrict results returned to those userswhoseemail value exactlymatches the specified email address.

n added cdrTagFilter, which if supplied, will restrict results returned to those userswhosecdrTag value exactlymatches the specified cdrTag.

Additional TURNserver types provided for /turnServers:

n expressway indicates theCisco ExpresswayX8.9 TURNserver is used.

n cms is equivalent to acanowhich is retained for legacy deployments.

2.9 Summary of CDR Additions & ChangesVersion 2.1 introduces the following changes to theCall Detail Records of theMeeting Server:

n support for up to 4 CDR receivers,

n new subType of distributionLink in thecallLegStart record, indicateswhen thecall leg is a conference distribution link to another Call Bridge in the cluster,

n new parameter replacesSipCallId in thecallLegStart record, and new reason forcall ending of callMoved in callLegEnd record.

2.10 Cisco endpoints no longer supportedFrom version 2.1, theMeeting Server is no longer tested for interoperabililtywith theseendpoints:

n Cisco TelePresenceSystem 3200 Series



n Cisco TelePresenceSystem 1000

n Cisco TelePresenceSystem 500-37 (onlywith 37 inch display)

as a consequenceMeeting Server 2.1 does not support the endpoints listed above, nor willrelated bugs be fixed in version 2.1.

Note: Version 2.0 of theMeeting Server will continue to support the endpoints listed above.



3 Notes on Installing and Upgrading to CiscoMeeting Server 2.1If you have recently purchased a Cisco Meeting Server 1000 or Acano X-series server, theMeeting Server software is already installed, however a new versionmay have recently beenreleased. Check the release using theMMP command version. If you are configuring a VM forthe first time then follow the instructions in theCisco Meeting Server Installation Guide forVirtualized Deployments. If your Meeting Server is running the latest software version then go toSection 3.2.

This section assumes that you are upgrading a Cisco Meeting Server 1000, an Acano X-Seriesserver or specified VM from 2.0.x. If you are upgrading from R1.9.x, then Cisco recommendsthat up you upgrade to 2.0.x first following the instructions in the 2.0.x release notes, beforefollowing any instructions in theseCisco Meeting Server 2.1 ReleaseNotes.

Note: It is possible to upgrade from release 1.8.x to Cisco Meeting Server 2.1 withoutupgrading to 1.9.x and 2.0.x, however this has not been tested byCisco.

3.1 Upgrading to Release 2.1The instructions in this section apply to bothMeeting Server and virtualized deploymentswith apreviousAcano server release already installed and not clustered. Refer to theScalability andResilienceDeployment Guide before upgrading clustered servers.

CAUTION: Before upgrading to release 2.1.0 youmust take a configuration backup using thebackup snapshot <filename> command and save the backup safely on a different device.See theMMPCommand Reference document for full details. Do NOT use the automatic backupfile that is created during the upgrade process.

Upgrading the firmware is a two-stage process: first, upload the upgraded firmware image;then issue the upgrade command. This restarts the server: the restart process interrupts allactive calls running on the server; therefore, this stage should be done at a suitable time so asnot to impact users − or users should bewarned in advance.

To install the latest firmware on the server follow these steps:

1. Obtain the appropriate upgrade file from the support section of theCisco website. Therewill be four files:

Cisco_Meeting_Server_2_1_0_vm-upgrade.zip

This file requires unzipping to a single upgrade.img file. Use this file to upgrade vm

3 Notes on Installing and Upgrading to Cisco Meeting Server 2.1


deployments, follow the instructions below.

Cisco_Meeting_Server_2_1_0.vhd

Use this file to upgradeMicrosoft Hyper-V deployments

Cisco_Meeting_Server_2_1_0_x-series.zip

This file requires unzipping to a single upgrade.img file. Use this file to upgradeAcano X-series servers, follow the instructions below.

Cisco_Meeting_Server_2_1_0.ova

Use this file for new vm deployments, follow the steps in the Installation Guide forVirtualized Deployments.

Note: If you are using WinSCP for the file transfer, ensure that the Transfer Settings optionis ‘binary’ not ‘text’. Using the incorrect setting results in the transferred file being slightlysmaller than the original – and this prevents successful upgrade.

2. Validate the download; the checksums for the 2.1.0 release are shown in a pop up box thatappearswhen you hover over the description for the download.

3. Using an SFTP client, log into theMMP using its IP address. The login credentialswill be theones set for theMMP admin account. If you are using Windows, we recommend using theWinSCP tool.

Note:a) You can find the IP address of theMMP’s interfacewith theiface aMMPcommand.b) TheSFTP server runs on the standard port, 22.c) After copying the upgrade.img file, youwill not be able to see it listed as being in the filesystem; this is normal.

4. Copy the software to theServer/ virtualized server.

5. To apply the upgrade, issue the upgrade command.

a. Establish a SSH connection to theMMP and log in.

b. Initiate the upgrade by executing the upgrade command.upgrade

TheServer/ virtualized server restarts automatically: allow 10minutes for the process tocomplete.



6. Verify that theMeeting Server is running the upgraded imageby re-establishing theSSHconnection to theMMP and typing:version

7. Check theConfiguration > Outbound Calls rules updating the Local Contact Domain fieldand completing the new Local From Domain field if necessary.

8. Update the customization archive filewhen available.

9. If you are deploying a scaled or resilient deployment read theScalability & ResilienceDeployment Guide and plan the rest of your deployment order and configuration.

10. If you have deployed a database cluster, be sure to run thedatabase cluster upgradeschema command after upgrading the database schema. For instructions on upgrading thedatabase schema refer to theScalability & ResilienceDeployment Guide.

11. You have completed the upgrade.

3.2 Cisco Meeting Server 2.1 DeploymentsTo simplify explaining how to deploy theMeeting Server, deployments are described in term ofthreemodels: the single combined Meeting Server, the single split Meeting Server and thedeployment for scalability and resilience. All three different modelsmaywell be used in differentparts of a production network.

3.2.1 Deployments using a single host server

If you are installing theMeeting Server for the first time on a single host server (a “combined”deployment), we recommend that you read and follow the documentation in the followingorder:

1. Appropriate Installation Guide for your Cisco Meeting Server (installation guide for CiscoMeeting Server 1000 and virtualized deployments or the installation guide for Acano X-Series Server).

2. TheSingleCombined Meeting Server Deployment Guide enabling all the solutioncomponents on the single host. This guide refers to theCertificateGuidelines for SingleCombined Server Deployments for details on obtaining and installing certificates for thisdeployment.

3.2.2 Deployments using a single split server hosted on a Core server and an Edgeserver

If you are installing theMeeting Server for the first time in a split server model, we recommendthat you deploy the XMPP server on theCore server, and deploy the Load Balancer on the Edgeserver.

Read and follow the documentation in the following order:



1. Appropriate Installation Guide for your Cisco Meeting Server

2. TheSingle Split Meeting Server Deployment Guide. This guide refers to theCertificateGuidelines for Single Split Server Deployments for details on obtaining and installingcertificates for this deployment.

3.2.3 Deployments for scalability and resilience

If you are installing theMeeting Server for scalability and resilience using multiple host servers,we recommend that you deploy the XMPP server on Core servers, and deploy Load Balancerson the Edge server.

Read and follow the documentation in the following order:

1. Appropriate Installation Guide for your Cisco Meeting Server

2. TheScalability and ResilienceDeployment Guide. This guide refers to theCertificateGuidelines for Scalable and Resilient Server Deployments for details on obtaining andinstalling certificates for this deployment.

3.3 DowngradingTo return to the previous version of the server software in a non-clustered environment, use theregular upgrade procedure to “upgrade” to the appropriate version. Then restore theconfiguration backup for the older version, using thebackup rollback <name> command.See theMMPCommand Reference document for full details. Do not rely on the backupgenerated automatically during upgrade.

Note: Thebackup rollback <name> command overwrites the existing configuration aswellas the license.dat file and all certificates and private keys on the system, and reboots theMeeting Server. Therefore it should be used with caution.Make sure you copy your existinglicense.dat file and certificates beforehand because theywill be overwritten during the backuprollback process. The .JSON filewill not be overwritten and does not need to be re-uploaded.



4 Resolved Issues

Resolved inMeeting Server 2.1.0

Reference Issue Summary

12044 SERVER-5143

Low frame rate sent toLync client after increasein window size to fullscreen

Low frame rate with high resolution sent to Lyncclient fromMeeting Server after Lync clientincreases window size to full screen. Fixed in 2.1.0

12005 SERVER-5131

Under heavy load,occasionally dynamicspaces are not created

Under heavy, sustained ad-hoc conference load,occasionally dynamic spaces are not handledproperly, leading to a failure of the ad-hocconference escalation. Fixed in 2.1.0

11943 CLIENT-

5322

In some situations thelogin page for the WebBridge may fail to load inyour browser

If the web browser attempts to supply a cookie of512 bytes or more to the Web Bridge, the loginpage will fail to load. A web browser may supplycookies belonging to other websites in the samedomain as the Web Bridge. Clearing your cookieswill resolve this issue. Fixed in 2.1.0

4 Resolved Issues


5 Known LimitationsThe following are known issues in this release. If you requiremore details on any of these pleasecontact Support, https://www.cisco.com/support.


- SERVER-5711

Active callsnot visiblefromWebAdmininterface

On the Status>Calls page of the Web Admin interface,the page is missing any active calls as well as the dropdown menus for status, configuration and logs that arenormally available.

- SERVER-5516

Participantlimits can beignored byload balancingfeature

Calls placed on a particular Call Bridge by the CallBridge Groups load balancing feature can ignore theparticipant limit set on that Call Bridge.

- SERVER-5519

Loadbalancing onlyapplies toincoming calls

Limit parameters on API node/system/configuration/cluster only apply to incomingcalls

11935 SERVER-5100

In a distributedconferencewith 3endpoints,there areissues withHold andResume

In a distributed conference, if there are only twoendpoints on one Call Bridge, if one holds and resumesand then the other leaves, the first will be shown thelobby screen as if it were the only endpoint in theconference.

11889 SERVER-5083

Heavyconferenceload can causeVM to crash

Meeting Server on VM can crash if under very heavy,sustained load involving small ad-hoc conferences.

11642 SERVER-4967

Heavyconferenceload can causeVM to dropcalls

Meeting Server on VM occasionally fails to reply to amid-call SIP reINVITE if under heavy, sustained ad-hocconference load, leading to call drops.

5 Known Limitations

https://www.cisco.com/support



11363 SERVER-4790

Lync 2013 andS4B 2016clients crashwhenreceivingcontent froman Acanodeployment

Pressing the ‘actual size’ button on Lync/S4B clientswhen receiving content from aMeeting Serverdeployment causes the Lync/S4B clients to crash.Experienced after upgrading to patch KB3115268 forS4B and patch KB3114944 for Lync 2013.

Workaround: use the dual homed meeting featurerather than calling the Lync/S4B clients directly into aspace.

11352 SERVER-4784

syslog may notwork for somecomponents

Enabling syslog for a Recorder, Streamer or Web Bridgewill not work (syslogs will not be written to the remotelocation) until the component in question has beenrestarted.

11058 SERVER-4646

Three screenTIP does notwork whendialing intoMeetingServer

When dialing a 3 Screen TIP endpoint into CiscoMeeting Server 2.0, it can fail to negotiate and drop to anormal single screen call. This happens if the SIP trunkfrom the CUCM to the Cisco Meeting Server isconfigured to use Early Offer.

Workaround: Set the SIP Trunk from the CUCM to theCisco Meeting Server to use Delayed Offer.

10611 SERVER-4411

One Recordercan onlyhandle up to 5simultaneousrecordings.

We recommend that each Recorder is used for amaximum of 5 simultaneous recordings.

9140 SERVER-3670

Endpointpresenceincorrectwhen alreadyin a Lyncmeeting

When an endpoint is dragged and dropped into a Lyncmeeting its presence is not correctly updated as busy.

8623 SERVER-3365

Noconferencecontrolpossible byAcano Clientof LyncClients,althoughcontrolsappear

When adding a space into a Lync conference withmultiple Lync users, an Acano app user can select aLync users name and conference control optionsappear (mute audio/video, remove) but these optionsdon't do anything.

5 Known Limitations

https://support.microsoft.com/en-us/kb/3115268

https://support.microsoft.com/en-gb/kb/3114944



8356 SERVER-3238

Syscall errorsin logs

If a WAN optimizer is deployed between clustereddatabase nodes, it may prevent keep-alive checks fromcompleting, causing SYSCALL errors to appear inlogs. In cases where aWAN optimizer is being usedbetween cluster nodes, it is important to ensure that allkeep alive traffic is sent in a timely manner.

Consult your WAN optimizer documentation on how toeither disable this functionality between specific IPaddresses, or for options that control whichoptimizations are applied.

5 Known Limitations


Cisco Legal InformationTHESPECIFICATIONS AND INFORMATIONREGARDING THEPRODUCTS INTHIS MANUAL ARESUBJECT TOCHANGEWITHOUT NOTICE. ALL STATEMENTS, INFORMATION, ANDRECOMMENDATIONS INTHIS MANUAL AREBELIEVED TOBEACCURATEBUT AREPRESENTEDWITHOUTWARRANTYOFANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKEFULL RESPONSIBILITY FORTHEIRAPPLICATIONOFANYPRODUCTS.

THESOFTWARE LICENSEAND LIMITEDWARRANTY FORTHEACCOMPANYING PRODUCTARESET FORTH INTHE INFORMATIONPACKET THAT SHIPPEDWITH THEPRODUCT ANDAREINCORPORATEDHEREINBY THIS REFERENCE. IF YOUAREUNABLE TO LOCATE THESOFTWARE LICENSEORLIMITEDWARRANTY, CONTACT YOURCISCOREPRESENTATIVEFORACOPY.

TheCisco implementation of TCP header compression is an adaptation of a programdeveloped by theUniversity of California, Berkeley (UCB) as part of UCB’s public domain versionof theUNIX operating system. All rights reserved. Copyright © 1981, Regents of theUniversityof California.

NOTWITHSTANDING ANYOTHERWARRANTYHEREIN, ALL DOCUMENT FILES ANDSOFTWAREOF THESESUPPLIERS AREPROVIDED “AS IS”WITHALL FAULTS. CISCOANDTHEABOVE-NAMEDSUPPLIERS DISCLAIM ALLWARRANTIES, EXPRESSEDOR IMPLIED,INCLUDING,WITHOUT LIMITATION, THOSEOFMERCHANTABILITY, FITNESS FORAPARTICULARPURPOSEANDNONINFRINGEMENT ORARISING FROM ACOURSEOFDEALING,USAGE, ORTRADEPRACTICE.

INNO EVENT SHALL CISCOOR ITS SUPPLIERS BE LIABLE FORANY INDIRECT, SPECIAL,CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING,WITHOUT LIMITATION, LOSTPROFITS ORLOSS ORDAMAGETODATA ARISING OUT OF THEUSEOR INABILITY TOUSETHIS MANUAL, EVEN IF CISCOOR ITS SUPPLIERS HAVEBEENADVISEDOF THEPOSSIBILITYOF SUCHDAMAGES.

Any Internet Protocol (IP) addresses and phone numbers used in this document are not intendedto be actual addresses and phone numbers. Any examples, command display output, networktopology diagrams, and other figures included in the document are shown for illustrativepurposes only. Any use of actual IP addresses or phone numbers in illustrative content isunintentional and coincidental.

All printed copies and duplicate soft copies are considered un-Controlled copies and theoriginal on-line version should be referred to for latest version.

Cisco hasmore than 200 officesworldwide. Addresses, phone numbers, and fax numbers arelisted on theCisco website at www.cisco.com/go/offices.

© 2017 Cisco Systems, Inc. All rights reserved.

Cisco Legal Information

http://www.cisco.com/go/offices


Cisco TrademarkCisco and theCisco logo are trademarks or registered trademarks of Cisco and/or its affiliatesin theU.S. and other countries. To view a list of Cisco trademarks, go to this url:www.cisco.com/go/trademarks. Third-party trademarksmentioned are the property of theirrespective owners. The use of theword partner does not imply a partnership relationshipbetweenCisco and any other company. (1110R)

Cisco Trademark

http://www.cisco.com/go/trademarks

Date post:	23-Jan-2021
Category:	Documents
Upload:	others
View:	32 times
Download:	0 times

Cisco Meeting Server 2.1.0 Release Notes...Release Notes,2.1 Release Notes, Cisco Meeting Server...

Documents