cisco meeting server 2.0 deployments with call control · rendezvous(alsoknown aspersonalcmr)...

CiscoMeeting ServerCisco Meeting Server Release 2.0Deployments with Call Control

November 18, 2016

Cisco Systems, Inc. www.cisco.com

http://www.cisco.com/

Cisco Meeting Server Release 2.0 : Deployments with Call Control Guide 2

Contents

1 Introduction 31.1 How to Use this Guide 3

1.1.1 Commands 51.1.2 Terminology 5

2 Configuring a SIP Trunk to Cisco Unified CommunicationsManager 62.1 Configuring a secure SIP trunk 6

2.1.1 Configuration required on theMeeting Server 62.1.2 Configuration required on theCisco Unified CommunicationsManager 8

2.2 Configuring a non-secure SIP trunk 102.2.1 Configuration required on theMeeting Server 102.2.2 Configuration required on theCisco Unified CommunicationsManager 10

3 Setting up scheduled and rendezvous calls 123.1 Configuring theMeeting Server 123.2 Configuring Cisco Unified CommunicationsManager 13

4 Setting up escalated ad hoc calls 154.1 Configuring theMeeting Server 154.2 Configuring Cisco Unified CommunicationsManager 15

A Configuring a SIP Trunk to an Avaya CM 16A.1 Configuration Summary 16A.2 Cisco Meeting Server Configuration 16A.3 Avaya CM Configuration 17

B Configuring a Polycom DMA for theCisco Meeting Server 22B.1 Setting up the External SIP Peer 22B.2 Creating theDial Rule 24

Cisco Legal Information 27


1 IntroductionTheCisco Meeting Server was formerly called theAcano Server. TheCisco Meeting Server isnow hosted on a new preconfigured version of a Cisco UCS server, called theCisco MeetingServer 1000. It can also be hosted on theAcano X-Series hardware, or on a specification basedVM server.

Note: The term Meeting Server in this document means either a Cisco Meeting Server 1000, anAcano X-Series Server or the software running on a virtual host.

This document provides examples of how to configure theMeeting Server to workwith CiscoUnified CommunicationsManager and other third party call control devices. The examplesmayneed to be adapted according to your specific deployment.

These instructions apply equally to all Meeting Server deployment topologies (single server andscaled/resilient deployments).

1.1 How to Use this GuideThis guide is part of the documentation set (shown in Figure 1) for theMeeting Server.

1 Introduction


Figure 1: Cisco Meeting Server documentation set

1 Introduction


1.1.1 Commands

In this document, commands are shown in black and must be entered as given - replacing anyparameters in <> bracketswith your appropriate values. Examples are shown in blue and mustbe adapted to your deployment.

1.1.2 Terminology

Throughout this document the conferencing typesmentioned are those as defined in Table 1.

Table 1: Conferencing Types

Conference type Description

Rendezvous (also knownas personal CMR)

Pre-defined, permanently available addresses that allow conferencing withoutprevious scheduling.

The host shares the address with other users, who can call in to that address at anytime.

Ad hoc Instant or escalated conferencing, for example manually escalated from a point-to-point call to a multiparty call with three or more participants.

Scheduled Pre-booked conferences with a start and end time.

1 Introduction


2 Configuring a SIP Trunk to Cisco UnifiedCommunications ManagerThis chapter explains how to set up a SIP Trunk betweenCisco Unified CommunicationsManager and aMeeting Server. TheMeeting Server must either be configured as a Core serverin a split server deployment, or as a single combined server.

Cisco recommends setting up a secure SIP trunk, however if your company policy is for trafficwithin your organization to be non secure, then a non-secure SIP trunk can be configured.However, the escalation of a 2-way call on Cisco Unified CommunicationsManager to aconference on theMeeting Server, requires theCisco Unified CommunicationsManager tocommunicatewith theAPI of theCisco Meeting Server. TheAPI requiresHTTPScommunication, so certificates need to be created and uploaded to both theCisco MeetingServer and Cisco Unified CommunicationsManager, and Cisco Unified CommunicationsManager needs to trust theMeeting Server's certificate, in order for escalated ad hoc calls towork.

If you are just allowing scheduled or rendezvous calls between theMeeting Server and CiscoUnified CommunicationsManager, and you set theSIP trunk as non-secure, then certificatesare not required. Definitions of call types are given in Section 1.1.2.

Note: If you are not your organization's Cisco Unified CommunicationsManager serveradministrator, then Cisco strongly advises you to seek the advice of your local administrator onthe best way to implement the equivalent on your server’s configuration.

2.1 Configuring a secure SIP trunkFollow the steps in Section 2.1.1 and Section 2.1.2 to set up the secure SIP trunk, then Chapter4 to enable escalation of a 2-way call on Cisco Unified CommunicationsManager to aconference on theMeeting Server.

2.1.1 Configuration required on theMeeting Server

Follow theCisco Meeting Server deployment guides to configure your Meeting Server, onceconfigured:

1. SSH into theMMPof theMeeting Server

2. If you have not already done so, specify a listening interface using theMMP commandcallbridge listen

3. Generate a private key and Certificate Signing Request (.csr) file for theCall Bridge, call it"cucm-trust.csr"

2 Configuring a SIP Trunk to Cisco Unified Communications Manager


Cisco Unified CommunicationsManager has some requirements onwhat TLS certificates itwill accept. You should ensure that "cucm.csr" has theSSL client and SSL server purposesenabled. This is done during the certificate signing stage.

4. Submit "cucm-trust.csr "to theCA (public CA or internal CA) for signing. An internal CAsigned certificate is acceptable.

5. Once signed, use openSSL to check that the certificate isOK:openssl x509 -in <certificatename> -noout -text –purposefor exampleopenssl x509 -in cucm-trust.crt -noout -text –purpose

The important lines in the output are "SSL client" and "SSL server" whichmust have a"Yes" against them, for example:

Certificate purposes:SSL client : YesSSL client CA : NoSSL server : Yes

6. Upload the signed certificate, and intermediateCA bundle (if any) to theCall Bridge

a. SSH into theMMP

b. Assign the certificate and private key to theCall Bridge using the command:

callbridge certs <keyfile> <certificatefile>[<cert-bundle>]

wherekeyfile and certificatefile are the filenames of thematching private keyand certificate. If your CA provides a certificate bundle then also include the bundle as aseparate file to the certificate.

For example:

callbridge certs cucm-trust.key cucm-trust.crt cucm-trust-bundle.crt

c. Restart theCall Bridge interface to apply the changes.

callbridge restart

If the certificate installs successfully on theCall Bridge, then the following is displayed:

SUCCESS: listen interface configuredSUCCESS: Key and certificate pair match

If the certificate fails to install, the following error message is displayed:

FAILURE: Key and certificate problem: certificate and key do notmatch

Note: Youwill need to add theCall Bridge certificate and certificate bundle to theCiscoUnified CommunicationsManager's trust store, see step 2 in section Section 2.1.2



Note: For more information on creating and uploading certificates to theMeeting Server,see the appropriateCisco Meeting Server CertificateGuidelines.

2.1.2 Configuration required on theCiscoUnified CommunicationsManager

Our testing has been done on trunkswithout Media Termination Point (MTP)configured. Therefore:

n DisableMTP if thiswill not negatively affect your deployment. Turning off MTPmight have anegative impact on your deployment if you are using SCCP phones and need to send DTMFto theMeeting Server.

n If the above is not a valid implementation, youmay need to increase theMTP capacity on theCisco Unified CommunicationsManager depending on the number of simultaneous calls.

1. Generate a certificate for Cisco Unified CommunicationsManager

a. Log into theCisco Unified CommunicationsManager OS Administration page

b. Select Security>CertificateManagement. TheCertificate List window displays.

c. Click theGenerateCSRbutton and generate a Certificate Signing Request(CSR) forCisco Unified CommunicationsManager.

d. Sign theCSRwith a CertificateAuthority. An internal CA signed certificate isacceptable.

e. Upload the signed certificate, private key, and intermediateCA bundle (if any) to CiscoUnified CommunicationsManager

2. Upload to theCisco Unified CommunicationsManager's trust store, the signed certificatecreated in step 4 of Section 2.1.1 for theMeeting Server's Call Bridge, and the rootcertificate or chain of certificates from theCertificateAuthority.

a. Log into theCisco Unified CommunicationsManager OS Administration page

b. Select Security>CertificateManagement.

c. Select Upload Certificate/CertificateChain.

d. Click Choose File to find your certificate. This can be the root certificate or theCallBridge’s certificate and certificate bundle.

e. Click Upload File.

3. Create a SIP trunk security profile

Cisco Unified CommunicationsManager applies a default security profile called Non SecureSIP Trunkwhen you create theSIP Trunk, this is for TCP. To use TLS, or something otherthan the standard security profile, follow these steps:



a. Log into Cisco Unified CommunicationsManager Administration.

b. Go to System > Security > SIP Trunk Security Profile.

c. Click Add New.

d. Complete the fields as follows:

l Name= type in a name, e.g. "CMS_SecureTrunk"

l DeviceSecurityMode=select Encrypted

l Incoming Transport Type = select TLS

l Outgoing Transport Type = select TLS

l X.509 Subject Name= enter theCNof theCall Bridge certificate.

l Incoming Port= enter the port whichwill receive TLS requests. The default for TLS is5061

e. Click Save

4. Create theSIP trunk

a. In Cisco Unified CommunicationsManager, go to Device >Trunk.

b. Click Add New.

c. Configure these fields:

l Trunk Type= SIP trunk

l DeviceProtocol =SIP

l Trunk Service Type= None (default)

d. Click Next

e. Configure the destination information for theSIP trunk, see Table 2 below.

Table 2: Destination information for the SIP Trunk

Field Description

Device name Type in a name e.g. CiscoMeetingServer (no spaces allowed)

Device pool The pool you want your device to belong to (as configured inSystem >Device Pool in Cisco Unified Communications Manager

SRTP Allowed Select SRTP Allowed to allowmedia encryption

Inbound Calls > Calling SearchSpace

Select default, not required if only allowing escalated 2-way adhoccalls from Cisco Unified Communications Manager to a meeting onthe Meeting Server.



Field Description

Outbound Calls > Calling PartyTransformation CSS

Select as appropriate.

SIP Information>Destinationaddress

Enter the FQDN of the Meeting Server, it must match the CN of theMeeting Server certificate

SIP Information>Destination Port Enter 5061 for TLS

SIP Trunk Security Profile Select the security profile that you created in step 3.

SIP Profile Select the Standard SIP Profile For TelePresence Conferencing

Normalization Script Assign cisco-telepresence-conductor-interop to this SIP trunk.Note: ideally download the latest normalization script from theCisco website. Even if you do not have a Conductor, the MeetingServer has the same interop issues that Conductor would have, andtherefore this script is suitable for a trunk to the core MeetingServer.

f. Click Save.

2.2 Configuring a non-secure SIP trunkFollow the steps in Section 2.2.1 and Section 2.2.2 to set up a non-secure SIP trunk, thenfollow Chapter 3 to enable rendezvous and scheduled calls betweenCisco UnifiedCommunicationsManager and Meeting Server.

2.2.1 Configuration required on theMeeting Server

Follow theCisco Meeting Server deployment guides to configure your Meeting Server, onceconfigured:

1. SSH into theMMPof theMeeting Server

2. If you have not already done so, specify a listening interface using theMMP commandcallbridge listen

2.2.2 Configuration required on theCiscoUnified CommunicationsManager

Our testing has been done on trunkswithout Media Termination Point (MTP)configured. Therefore:

n DisableMTP if thiswill not negatively affect your deployment. Turning off MTPmight have anegative impact on your deployment if you are using SCCP phones and need to send DTMFto theMeeting Server.

n If the above is not a valid implementation, youmay need to increase theMTP capacity on theCisco Unified CommunicationsManager depending on the number of simultaneous calls.



1. Create theSIP trunk

2. a. In Cisco Unified CommunicationsManager, go to Device >Trunk.

b. Click Add New.

c. Configure these fields:

l Trunk Type= SIP trunk

l DeviceProtocol =SIP

l Trunk Service Type= None (default)

d. Click Next

e. Configure the destination information for theSIP trunk, see Table 2 below.

Table 3: Destination information for the SIP Trunk

Field Description

Device name Type in a name e.g. CiscoMeetingServer (no spaces allowed)

Device pool The pool you want your device to belong to (as configured inSystem >Device Pool in Cisco Unified Communications Manager

SRTP Allowed Do not allow SRTP

Inbound Calls > Calling SearchSpace

Select default.

Outbound Calls > Calling PartyTransformation CSS

Select as appropriate.

SIP Information>Destinationaddress

Enter the destination address e.g.ciscomeetingserver.example.com or an IP address

SIP Information>Destination Port Enter 5060 for TCP

SIP Trunk Security Profile Not required for a non-secure SIP trunk

SIP Profile Select the Standard SIP Profile For TelePresence Conferencing

Normalization Script Not required for a non-secure SIP trunk.

f. Click Save.



3 Setting up scheduled and rendezvous callsAfter setting up a secure SIP trunk (Section 2.1 ) or non-secure SIP trunk (seeSection 2.2),follow the steps in Section 3.1 and Section 3.2 to enable rendezvous and scheduled calls to bemade from theMeeting Server to Cisco Unified CommunicationsManager.

3.1 Configuring the Meeting Server1. Configure an outbound dial plan rule for calls that will be sent to Cisco Unified

CommunicationsManager from theMeeting Server.

a. Using theWeb Admin Interface of theMeeting Server, go to Configuration > OutboundCalls.

Table 4: Outbound Calls

b. In the blank row, for Domain, enter the domain that will bematched for calls that need tobe sent to Cisco Unified CommunicationsManager.

c. For SIP proxy to use, do one of the following:

l Leave this field blank and the server will perform a DNS SRV lookup for the calleddomain using _sip.tls.<yourcucmdomain>.com. If this fails to resolve, theMeeting Server will try a lookup using TCP and then UDP.

or

l Enter theCUCM FQDN, the server will perform a DNS SRV lookup for that defineddomain.

Note: If this fails to resolve, theMeeting Server will try a lookup using TCP and thenUDP. The server will then perform a DNS A record lookup for theHost entered, if theaboveSRV lookup fails to resolve using TLS, TCP or UDP.

or

l Enter the IP address of your Cisco Unified CommunicationsManager.

d. For Local contact domain, leave this field blank, it is only required when setting up a SIPtrunk to Lync.

3 Setting up scheduled and rendezvous calls


e. For Local from domain, enter the domain that youwant the call to be seen as comingfrom (theCaller ID).

Note: If you leave Local from domain blank, the domain used for theCaller ID defaults tothat entered in the Local contact domain, in this case blank.

f. For Trunk type select Standard SIP.

g. Set the Priority as required.

h. Select Add New.

3.2 Configuring Cisco Unified CommunicationsManager1. Configure the dial plan for outbound calls

You can configure number based routing e.g. 7xxx, or domain based routing [email protected], to theMeeting Server. In both cases this is done through theCisco Unified CommunicationsManager interface. Follow the relevant example here:

Domain based routing example

To route all domain based calls from Cisco Unified CommunicationsManager to theMeeting Server then:

a. Log into theCisco Unified CommunicationsManager Administration interface:

b. Go to Call Routing > Sip Route Pattern.

c. Click Add New.

d. Complete the following:

l Pattern usage= domain routing

l IPv4 pattern something likemydomain.example.com

l Description = anything youwant

l Route partition = see below

l SIP trunk / route list = the trunk you have already configured

e. Click Save.

Numeric dialing example

This basic example routes everything starting with a 7 to theMeeting Server.

a. Log into theCisco Unified CommunicationsManager Administration interface:

b. Go to Call routing > Route/Hunt > Route Pattern.

c. Click Add New.



d. Complete the following:

l Route pattern = 7.! (The ! means anything. The dot is useful for a later option below.)

l Route partition = the route partition youwant this rule to belong to - see the notebelow

l Description = any appropriate text

l Gateway/route list = the trunk you have already configured

l Route this pattern = ensure that this option is selected

Further down the page you can set various transforms e.g. in theDiscard Digits fieldyou can select PreDot to strip off the leading 7 in our example.

e. Click Save.

Note: : Various dial plan rules are attached to a route partition and a calling searchspace (CSS) comprises a list of route partitions. You can have a different CSS fordifferent people, each phone, or trunk.When a call ismade, Cisco UnifiedCommunicationsManager goes through each route partition in theCSS until it findsone that has a matching rule.

2. Test.

Make some test calls



4 Setting up escalated ad hoc callsAfter setting up the secure SIP trunk (seeSection 2.1), follow the steps in Section 4.1 andSection 4.2 to enable the escalation of a 2-way call on Cisco Unified CommunicationsManagerto a conference on theMeeting Server.

Note: I f you decided to set up theSIP trunk as non-secure, youwill still need to use certificates,as the escalation of a 2-way call on Cisco Unified CommunicationsManager to a conference ontheMeeting Server, requires theCisco Unified CommunicationsManager to communicatewiththeAPI of theCisco Meeting Server. TheAPI requiresHTTPS communication, so certificatesneed to be created and uploaded to both theCisco Meeting Server and Cisco UnifiedCommunicationsManager and each needs to trust the other's certificate, in order for escalatedad hoc calls to work.

4.1 Configuring the Meeting Server1. Set up an incoming dial plan on theMeeting Server see theCisco Meeting Server

Deployment Guide. For adhoc calls the rule should match against spaces.

2. Set up an administrator user account with "api" permission for Cisco UnifiedCommunicationsManager to use. See theCisco Meeting Server MMPCommand LineReferenceGuide.

4.2 Configuring Cisco Unified CommunicationsManager1. Create theConferenceBridge

a. In Cisco Unified CommunicationsManager Administration, select Media Resources >ConferenceBridge. The Find and List ConferenceBridgeswindow displays.

b. Click Add New. TheConferenceBridgeConfigurationwindow displays.

c. Select Cisco TelePresenceConductor from theConferenceBridge Typedrop-downlist.

d. Enter a nameand description for theMeeting Server in theDevice Information pane.

e. Select a SIP trunk from theSIP Trunk drop-down list.

f. Enter theHTTP interface information and check theHTTPS check box to create asecureHTTPS connection betweenCisco Unified CommunicationsManager andCisco Meeting Server.

2. If not already done so (see step 2 in Section 2.1.2), upload theCall Bridge's certificate andkey pair to theCisco Unified CommunicationsManager's trust store.

4 Setting up escalated ad hoc calls


A Configuring a SIP Trunk to an Avaya CMThis appendix provides an example of setting up a SIP trunk between theCisco Meeting Serverand theAvaya CommunicationsManager (Avaya CM) and may need to be adapted.

Note: If you are not your organization's Avaya CM administrator, then Cisco strongly advises youto seek the advice of your local administrator on the best way to implement the equivalent onyour server’s configuration.

Note: Avaya CM is an Avaya PBX, so callswill be audio only, however, theCisco Meeting Serverdoes not impose this restriction on interoperabilitywith Avaya: therefore a call defined to betype ‘avaya’ in theMeeting Server does not imply that the call is audio-only.

A.1 Configuration SummaryThis example deployment assumes that:

n This audio connection between Avaya CM and theMeeting Server is accessed via dialing aprefix 49

n The assigned IVR digits for theMeeting Server are 8320; that is a user from theAvayaenvironment will dial 498320 to access theMeeting Server IVR

n ADID extension 5328 to route to this samenumber and allow for PSTNdial-in to theMeetingServer

n Avaya Software Version: CM6 R016x.00.1.510.1 Update: 19940

A.2 Cisco Meeting Server Configuration1. Log in to theWeb Admin Interface and go to Configuration > General.

2. For IVRNumeric ID, enter 8320.

These digitswill be passed from theAvaya CM to theMeeting Server, and then routed to theMeeting Server IVR.

3. Click Submit.

4. Go to Configuration > Outbound Calls.

A Configuring a SIP Trunk to an Avaya CM


5. Add a dial plan entry for theAvaya CM – see the example below.

The highlighted IP address below matches theC-LANor Processor Ethernet address on theCM side and represents theCM interface used in theSignaling Group created later.

6. Click Add New.

A.3 Avaya CMConfiguration1. Add a node name for theMeeting Server signaling interface.

2. Add an Avaya Signaling Group with the following:

l Group Type= SIP

l Near-end NodeName= C-LANor Processor Ethernet interface indicated in the dial plansetting in the previous section

l Far-end NodeName= Node name for theMeeting Server signaling interface createdabove.

l Port settings for both Near-end and Far-end = 5060

l Far-end Domain = SIP domain associated with theMeeting Server

l Direct IP-IP Audio Connections = n. This ensures that all traffic from theAvaya CM comesfrom theNear-end Node



3. Add an Avaya TrunkGroup with the following:

l Group Type= SIP

l Direction = two way

l Service Type= tie

l Additional settingsmay vary, but see the examples below for possible configuration



4. Add an Avaya Route Pattern to routes calls to trunk group 105 and delete the first two digits(deletes the prefix digits 49).



5. Add a Uniform Dial Plan to provide a routing for a 6-digit number with a prefix of 49. Thesecallsmust be set to be routed to AAR tables in Avaya.

6. Add an AAR setting to routes all calls of 6 digits in length and beginning with 49 (i.e.498320) to route pattern 105 (theMeeting Server TrunkGroup).

7. Assign an Extension and DID.

Optionally, in theUniform Dial Plan you can add a setting for a DID extension (in this example,x5328) to route a call via digits498320 to theCisco Systems server.



B Configuring a Polycom DMA for the CiscoMeeting ServerFor calls from a Polycom DMA environment to theCisco Meeting Server, create an External SIPPeer on the Polycom DMA that will point to theMeeting Server, and then configure a Dial Rule onthe Polycom DMA that will direct calls to it.

The following is an example of configuring theMeeting Server for the Polycom DMA, and mayneed to be adapted. Follow the instructions in theDeployment guides to set up a dial plan rulethat points to the Polycom DMA server in theWeb Admin InterfaceConfiguration > OutboundCalls page. Also ensure that the correct ports are open (Incoming/Outgoing UDP 32768-65535 – RTP).

Note: If you are not your organization's Polycom server administrator, then Cisco stronglyadvises you to seek the advice of your local administrator on the best way to implement theequivalent on your server’s configuration.

B.1 Setting up the External SIP PeerOn thePolycom DMA:

1. Go to Network > External SIP Peer > Add

2. In the External SIP Peer page configure the following:

l Name: Cisco Systems

l Description: a meaningful phrase, possibly Cisco Systems IP Peer

l Next hop Address: IP Address of theMeeting Server Call Bridge

B Configuring a PolycomDMA for the Cisco Meeting Server


l Port: 5060

l User RouteHeader: selected

l Type: Other

l Transport Type: TCP

3. Leave theDomain List page blank.

4. In the Postliminary pageHeader Options section configure the following:

a. CopyAll Parameters: Checked

b. Format: Use original request's To

5. In the Postliminary pageRequest URI options section configure the following:

a. Format: Original user, configured peer's Destination Network or next hop address



6. In theAuthentication page configure the following:

a. Authentication: Pass authentication

b. Proxy authentication: Pass Proxy authentication

7. Click Save.

B.2 Creating the Dial RuleIn the Polycom DMA:



1. Go to Admin > Call Server > Dial Rules > Add.

2. In the Edit Dial Rule for Authorized Calls page, configure the following (see below):

a. Description: Cisco <Description of pattern>

3. Select Enabled.

4. Select theCisco SystemsSIP Peer in the left pane and click the arrow to move it to theSelected SIP Peers.

5. In the Preliminary page create a string to represent how callswill match this rule (see below).



Consult theDMAAdmin Guide for more detail. The example below matches any call thatbeginswith a 6 and sends it to the [[[Undefined variable BrandingTypeVariables.solution orserver]]].

if(!DIAL_STRING.match(/sip:6/)){return NEXT_RULE;}

6. ClickOK.

You should now be able to dial from anySIP-enabled Polycom DMA endpoint to theCiscoMeeting Server using the rule created.



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.

© 2016 Cisco Systems, Inc. All rights reserved.

THESPECIFICATIONS AND INFORMATIONREGARDING THEPRODUCTS INTHIS MANUAL ARESUBJECT TOCHANGEWITHOUT NOTICE. ALL STATEMENTS, INFORMATION, ANDRECOMMENDATIONS INTHIS MANUAL AREBELIEVED TOBEACCURATEBUT ARE

Cisco Legal Information

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


PRESENTEDWITHOUTWARRANTYOFANY 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.

The following information is for FCC compliance of Class A devices: This equipment has beentested and found to complywith the limits for a Class A digital device, pursuant to part 15 of theFCC rules. These limits are designed to provide reasonable protection against harmfulinterferencewhen the equipment is operated in a commercial environment. This equipmentgenerates, uses, and can radiate radio-frequency energy and, if not installed and used inaccordancewith the instructionmanual, may cause harmful interference to radiocommunications. Operation of this equipment in a residential area is likely to cause harmfulinterference, in which case userswill be required to correct the interference at their ownexpense.

The following information is for FCC compliance of Class Bdevices: This equipment has beentested and found to complywith the limits for a Class Bdigital device, pursuant to part 15 of theFCC rules. These limits are designed to provide reasonable protection against harmfulinterference in a residential installation. This equipment generates, uses and can radiate radiofrequency energy and, if not installed and used in accordancewith the instructions,may causeharmful interference to radio communications. However, there is no guarantee that interferencewill not occur in a particular installation. If the equipment causes interference to radio ortelevision reception, which can bedetermined by turning the equipment off and on, users areencouraged to try to correct the interference by using one ormore of the following measures:

l Reorient or relocate the receiving antenna.

l Increase the separation between the equipment and receiver.

l Connect the equipment into an outlet on a circuit different from that to which the receiveris connected.

l Consult the dealer or an experienced radio/TV technician for help.

Modifications to this product not authorized byCisco could void the FCC approval and negateyour authority to operate the product.

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 theUniversity of 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.

© 2016 Cisco Systems, Inc. All rights reserved.


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