Cisco UC Microsoft

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of 72 EDCS# 861590 Initial version

Application Note

Microsoft Office Communications Server 2007 R2 Enterprise Edition using Direct SIP to Cisco Unified Communications Manager 7.1(3)

Table of ContentsTable of ContentsTable of ContentsTable of Contents

Introduction..............................................................................................................................................................................................................2 Network Topology....................................................................................................................................................................................................3 Limitations................................................................................................................................................................................................................4 Other Known Limitations not tested in this setup.....................................................................................................................................................5 System Components.................................................................................................................................................................................................6

Hardware Requirements......................................................................................................................................................................................6 Software Requirements........................................................................................................................................................................................6 Features Supported..............................................................................................................................................................................................7 Features Not Supported.......................................................................................................................................................................................7

Configuration............................................................................................................................................................................................................8 Configuring Microsoft OCS 2007 R2 Enterprise Edition ...................................................................................................................................9 Configuring Cisco Unified Communications Manager.....................................................................................................................................43 Configuring SIP Gateway..................................................................................................................................................................................63

Acronyms................................................................................................................................................................................................................70

© 2010 Cisco Systems, Inc. All rights reserved.

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IntroductionIntroductionIntroductionIntroduction

• This document describes Direct SIP interoperability and documents the steps and configurations necessary for Cisco Unified Communications Manager (Cisco UCM) Release 7.1(3) to interoperate with Microsoft Office Communications Server (OCS) 2007 R2 Enterprise Edition. Testing has been done according to Microsoft’s Open Interoperability (OIP) Direct SIP program test plan, which includes end-to-end testing as well as testing with Microsoft’s Open Interoperability Test Tool (OITT) used to simulate the OCS 2007 R2 environment. The Microsoft test plan covers scenarios to test Gateways/PBX’s or IP-PBX’s, per Microsoft’s Direct SIP specifications. The application note aims to provide a good understanding of what works and what does not work in terms of the interaction between a Cisco UCM device and Microsoft Office Communicator (OC). It also provides guidance to deployment participants regarding the limitations, expected behaviors, and known issues. Please note that this document does not address performance and scalability, which are part of a broader criteria for a deployment-ready solution (for more details refer to 7.0 SRND: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/uc7_0.html).

• The endpoints used in this testing all have E.164 numbering which is supported by Cisco UCM Release 7.x.

• An alternative to Direct SIP, which provides only basic SIP trunk interoperability and requires dual call control systems, is Cisco UC Integration for Microsoft Office Communicator. This tight integration for Microsoft Office Communicator gives you the benefits of investment protection and reduced complexity delivered by a single, proven call control solution. It takes advantage of a common unified client services framework, so you gain:

a. Increased productivity, to instantly connect with colleagues, partners, and customers from anywhere and have a wide-band audio and high-definition video communications experience with the integrated Cisco IP softphone

b. Streamlined communications, with telephony presence, visual voicemail, communications history, and Cisco Unified IP Phone control from your desktop

c. Enhanced collaboration, allowing you to initiate or escalate into integrated voice, video, and web Cisco Unified MeetingPlace and Cisco WebEx sessions with multiple parties

d. Reduced complexity, through an easy-to-deploy integration and a single-call control architecture

e. Investment protection, so you can make an immediate business impact with Cisco Unified Communications, while protecting investments in existing desktop applications

http://www.cisco.com/en/US/products/ps10317/index.html

http://www.cisco.com/en/US/products/ps10317/index.html



Network TopologyNetwork TopologyNetwork TopologyNetwork Topology

Figure 1. Network Topology/Test Setup for End to End Test scenarios

Laptop

Laptop

LaptopLaptop Laptop



LimitationsLimitationsLimitationsLimitations

• OCS 2007 R2 does not support calling and connected name. Cisco UCM sends its calling name and number to mediation server/OCS, but the OC only displays the calling number. Name is not sent by Mediation Server.

• OCS 2007 R2 cannot be configured to restrict calling name and number.

• OCS 2007 R2 does not provide music on hold (MoH) capability except on Attendant Console.

• OCS 2007 R2 does not support Encryption TLS-SRTP on SIP trunk to CUCM.

• OCS 2007 R2 might not be able to handle port change for mid-call reINVITEs.

• OCS 2007 R2 displays presence information in the OC only for other OCS users configured in the Active Directory and OCS enabled.

• On OCS 2007 R2, the OC cannot be configured for Call Forward Busy. As an option the OC application allows the user to receive a notification of an incoming call during an active call and “redirect” the incoming call to a destination of the user’s choice.

• When OCS 2007 R2 initiates a conference call, and the call ends, the conference session is not dropped until all the parties hang up including the initiator.

• For OCS 2007 R2 to Cisco UCM inter-working, media termination point is required on the SIP trunk for proper operation.

• OCS 2007 R2 implementation of Early Media does not follow the standard implementation of the RFC used by the majority of PBX’s in the field. As a result Cisco UCM behavior might be different from what some early media test scenarios from the Microsoft test plan expect.

• OCS 2007 R2 only supports TCP transport to carry SIP messages. If UDP is required CUBE can be used to perform the conversion.

• OCS 2007 R2 only supports G711 ulaw or alaw on the outside interface. If G729 is required CUBE can perform the transcoding function.

• During a three-way conference initiated by an OC user the OC user is unable to “mute” any participant that is connected to the conference using a Cisco Unified IP phone. The issue is caused by the incompatibility between Microsoft OCS R2 media keepalive mechanism against Cisco Unified Communications solutions. Microsoft OCS R2 utilizes a Re-INVITE session to establish single direction media toward the muted user (in the case of OC MoH is not played but media is still forced to “sendonly-recvonly” attributes). During the time that the muted end-point remains in recvonly mode, Microsoft OCS system expects to receive RTCP packets from the muted end-point as a means to detect the muted user has not dropped from the conference. Cisco Unified Communications does not support RTCP in recvonly mode which causes the call to be dropped, after 30secs, from the Microsoft OCS side (Note: Cisco Unified Communications utilizes an ICMP solution to monitor livehood of an end-point on hold). Performing a “hold” from the OC side does not cause this issue, performing a hold forces a “inactive” media state and no RTCP packet is expected by the Microsoft mediation server. The issue described also applies to the Microsoft Attendant Client for hold (Attendant client supports MoH during a hold, which forces a single direction media negotiation) and all other Microsoft OCS applications that support MoH (single media direction) toward a Cisco Unified IP phone.

• Cisco UCM responds with a 404 and Q.850, cause = 1 (Unallocated number) when T1 line is disconnected (when MGCP Gateway is used instead of SIP Gateway, UCM will respond with 404 and Q.850, cause = 27 (Destination out of order)). This is because the T1 line disconnect causes the gateway ports to unregister on UCM and any routes to these ports are no longer allocated.

• Cisco UCM does not support comfort noise negotiation.

• Cisco UCM does not support mid-call codec/IP/port update without inactive in between.

• Cisco UCM SIP trunk can only be set to one codec at a time.

• If Cisco UCM receives a SIP INVITE with E.164 DN (containing a “+”) and “ext” attribute, and UCM has a route pattern for that E.164 DN out to the PSTN/GW, then UCM will send a 404 Not Found and will not route the call. This occurs because when the “ext” attribute is present in the INVITE, the "+" from the DN gets stripped and so UCM is not able to match the route pattern to the number received/dialed. Under the same scenario, this issue does not occur if the DN is non-E.164 (so route pattern does not have a “+”) or if E.164 DN is that of an endpoint registered to the Cisco UCM (so no route pattern is needed). This is a known issue on Cisco UCM 7.1(3) and has been documented in CDETS CSCtf66290 to be fixed in a future release. This is not an issue if the “ext” attribute is not sent in the SIP INVITE.



Other Known Limitations not tested in this setupOther Known Limitations not tested in this setupOther Known Limitations not tested in this setupOther Known Limitations not tested in this setup

• The Microsoft Mediation Server does not support video transcoding, thus video is not possible between OCS and UCM endpoints.

• Call Admission Control (CAC) is available for UCM endpoints but not OCS endpoints.

• Survivable Remote Site Telephony (SRST) is available for UCM endpoints but not OCS endpoints.

• E.911 support is available for UCM endpoints but not OCS endpoints.



System ComponentsSystem ComponentsSystem ComponentsSystem Components

Hardware Requirements

• Cisco MCS 7825H servers

• Cisco Unified IOS gateway 3825 (with VIC 4FXS/DID, VIC2-4FXO, VWIC2-2MFT-T1/E1 modules)

• Cisco Catalyst 3560

• 1 Cisco 7975 IP Phone (SCCP)

• 1 Cisco 7971 IP Phone (SIP)



• DELL notebook computers running Windows XP Professional version 2002 with SP2 (for OC)

Software Requirements

• Cisco Unified Communications Manager Release 7.1(3)

• Microsoft Office Communications Server 2007 R2 Enterprise Edition, Windows Server 2008 x64 Enterprise Edition OS

• Windows Active Directory/DNS/Cert Server Enterprise Edition for OCS 2007 R2, Windows Server 2008 x64 Enterprise Edition OS

• Windows SQL Server 2008 Enterprise Edition, Windows Server 2008 x64 Enterprise Edition OS

• Microsoft Mediation Server for OCS 2007 R2 EE, Windows Server 2008 x64 Enterprise Edition OS

• Microsoft Office Communicator 2007 R2 version

IMPORTANT: Microsoft Office Communications Server, Mediation Server and Office Communicator all need to run through the Microsoft Update website to have the latest updates installed for this integration to work correctly.



Features Supported

• Basic calls (G711 u-law and A-law)

• Conference calls

• DTMF

• PRACK

• Early Media

• Call hold and resume

• SDP Parsing

• Session Renegotiation

• ISDN Mapping

Features Not Supported

• Encryption (TLS-SRTP)



ConfigurationConfigurationConfigurationConfiguration

The integration of Cisco UCM to Microsoft OCS 2007 R2 is very similar to that of Microsoft OCS 2007. The differences, if any, might involve

the look and feel of some of the configuration windows (refer to the captured screenshots), however the information entered is still the same.

The main differences were seen during the deployment of the OCS 2007 R2 environment itself and how its own components connected to each

other. Those differences include, the use of 64-bit servers for each component (refer to Software Requirements section), the need for an

interface module (automatically downloaded during installation) on the Front End Servers to enable communication with SQL 2008 (refer to the

Microsoft deployment and installation guides for more information, links included below), and finally the configuration of the certificate

authority server (refer to the Microsoft deployment and installation guides for more information, links included below).

For Cisco deployment and installation guides, go to:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/uc7_0.html

For Microsoft deployment and installation guides, go to:

http://social.technet.microsoft.com/Search/en-

US/?query=office%20communications%202007%20r2%20server&ac=8

http://www.microsoft.com/downloads/details.aspx?familyid=F19AE8E0-311C-4CE1-BBA2-

5702F32F5409&displaylang=en

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/uc7_0.html

http://social.technet.microsoft.com/Search/en-US/?query=office%20communications%202007%20r2%20server&ac=8

http://social.technet.microsoft.com/Search/en-US/?query=office%20communications%202007%20r2%20server&ac=8

http://www.microsoft.com/downloads/details.aspx?familyid=F19AE8E0-311C-4CE1-BBA2-5702F32F5409&displaylang=en

http://www.microsoft.com/downloads/details.aspx?familyid=F19AE8E0-311C-4CE1-BBA2-5702F32F5409&displaylang=en



Configuring Microsoft OCS 2007 R2 Enterprise Edition

Please refer to the Microsoft OCS 2007 R2 Enterprise Edition deployment guide for setup details. Only interoperability related information is included in this document.

Domain Name System Configuration

Forward Lookup Zone

Host Records added for the Cisco UCM and OCS Front End pools.



Reverse Lookup Zone

PTR Records added for the OCS Front End pools.



PTR Records added for the Cisco UCM.



SRV records

SRV Records added for the OCS Front End pools.



Front End Server/Pool Configuration

Pool Properties

Created a location profile with a (set of) normalization rule(s) for the OCS pools. The normalization rule(s) apply the chosen translation pattern to dialed numbers.



Click to add a location profile.



Click to add normalization rule(s).



Normalization rule associated with the location profile for the OCS environment.



Mediation Server Configuration



The Mediation Server has two Ethernet Interfaces, one listens to the OCS on 172.21.1.200 and the other to the Cisco UCM on 172.20.155.50.



FQDN for pool1 of OCS.

IP Address of Cisco UCM.



Make sure a valid certificate is assigned to the mediation server.



Added a route for the number patterns to be routed through this Mediation Server inbound or outbound.



Meditation Server Properties

Completed configuration.



User Configuration



After adding the users in the Active Directory, right-click any user and choose “Properties” to go through the configuration.



Configure the telephony settings.



Pool Users



Microsoft Office Communicator 2007 R2 Configuration



Enter the user’s sign-in address, click the Advanced button to select the Advanced Connection Settings.



This needs to be configured for client-to-server SIP communication when no SRV record for the SIP service provider is created in the DNS.



Sign in to the Microsoft Office Communicator 2007 R2.



Sign-in complete.



Configuring Cisco Unified Communications Manager

Cisco Unified Communications Manager Version



SIP Trunk

From the “Cisco Unified CM Administration” page, go to “Device” then click “Trunk”.

Add a SIP Trunk connecting to the Microsoft Mediation Server. Check the box for “Media Termination Point Required” to ensure complete inter-working with the Mediation Server.

Configure and assign the Media Resource Group List to the SIP trunk to ensure MTP Software and Conferencing resources are available.

SIP Trunk connecting to the Microsoft Mediation Server.

See “Media Resource Group Configuration” section.

To ensure complete inter-working with Microsoft Mediation Server.



For “Inbound Calls” select “All” to accept all number patterns incoming into the Cisco UCM from Microsoft Mediation Server. Under “SIP Information” make sure “Destination Address” contains the IP Address of the Microsoft Mediation Server.

All number patterns are accepted into the Cisco UCM from Microsoft Mediation Server..

IP Address of Microsoft Mediation Server..



Add a SIP Trunk connecting to the OITT. Check the box for “Media Termination Point Required” to ensure complete inter-working with OITT.

Configure and assign the Media Resource Group List to the SIP trunk to ensure MTP Software and Conferencing resources are available.

SIP Trunk connecting to the OITT.

To ensure complete inter-working with OITT.

See “Media Resource Group Configuration” section.



For “Inbound Calls” select “All” to accept all number patterns incoming into the Cisco UCM from OITT. Under “SIP Information” make sure “Destination Address” contains the IP Address of the OITT.

All number patterns are accepted into the Cisco UCM from OITT.

IP Address of OITT.



Add SIP trunk to SIP Gateway for PSTN calls.

SIP Trunk connecting to the SIP Gateway.



All number patterns are accepted.

IP Address of SIP Gateway.



Media Resources Group Configuration

From the “Cisco Unified CM Administration” page, click “Media Resources” and choose “Media Resources Group” to create a new group.

Click “Media Resources” and choose “Media Resources Group List” to create a new list and assign the new group to it.

Choose all needed resources.

Assign corresponding Media Resource Group.



Cisco IP Phone Configuration

From the “Cisco Unified CM Administration” page, go to “Device” then click “Phone”.

Add corresponding IP Phones. Shown below are the IP phones that were configured for this setup.



7970 SIP Phone

Ensure fields in red rectangles below are configured.



On the “Phone Configuration” page under “Association Information”, click “Line [1] – Add a new DN” to enter the “Directory Number Configuration”. The red rectangles below highlight the fields to configure and/or check.



Route Pattern Configuration

From the “Cisco Unified CM Administration” page, choose “Call Routing” � “Route Hunt” � “Route Pattern”.

Route pattern setup to route calls from Cisco UCM to the Microsoft OCS R2.

Route pattern to OCS R2 endpoints.

SIP trunk to Microsoft Mediation Server.

Prefix with a plus sign (+) to match the directory numbers (DN’s) on the OCS R2 side, in case the normalization rules in the Microsoft OCS R2 environment do not already add the plus sign.



Setup a route pattern to route calls to the OITT environment. Prefix with the corresponding digits to match the users directory numbers (DN’s) configured on the OITT.

Route to SIP trunk for OITT.

Route pattern to OITT.

Prefix digits to match the user directory numbers (DN’s) configured on the OITT.



Setup a route pattern to route certain numbers out to the PBX/PSTN.

Route pattern for dialing to PBX/PSTN numbers.

Route to SIP Gateway to PBX/PSTN.



Translation Pattern Configuration

From the “Cisco Unified CM Administration” page, click “Call Routing” then choose “Translation Pattern”.

Add a Translation Pattern to prefix digits to the Calling Party dialing to OITT to make it appear as an E.164 number.



Add a Translation Pattern to strip the + from numbers dialed from OITT going out to the PBX/PSTN.



Configuring SIP Gateway

Add a SIP Gateway connection to Cisco UCM to connect to PBX/PSTN.

! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname MS_GW1 ! boot-start-marker boot system flash c3825-ipvoicek9-mz.124-22.T3.bin boot-end-marker ! card type t1 0 1 logging message-counter syslog logging buffered 10000000 enable secret 5 $1$rqA7$jIYZ9c.TQOjnATrchIPN40 enable password CSCO123 ! no aaa new-model network-clock-participate wic 1 network-clock-select 1 T1 0/1/0 ! dot11 syslog ip source-route ip cef ! ! ! ! ip host CM-KINGS 172.20.150.251 no ipv6 cef multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! voice service pots ! voice service voip sip session transport tcp ! ! voice class codec 1



codec preference 1 g711ulaw codec preference 2 g711alaw ! ! ! ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /^\+1/ /1/ ! voice translation-rule 100 rule 1 /^$555$+$.*$/ /444\2/ rule 2 /^$12$+$.*$/ /911\2/ ! voice translation-rule 200 rule 1 /^$12$+$.*$/ /911\2/ ! voice translation-rule 400 ! ! voice translation-profile RemovePlus translate called 1 ! voice translation-profile test translate called 1 ! ! ! ! ! ! username cisco privilege 15 secret 5 $1$g9MP$FSHytQTj7RrCjPsHV5.tL/ archive log config hidekeys ! ! controller T1 0/1/0 cablelength long 0db pri-group timeslots 1-3,21-24 ! controller T1 0/1/1 cablelength long 0db pri-group timeslots 1-24 service mgcp ! ! class-map match-all voip match ip dscp ef class-map match-any telepresence



description CiscoTelepresence match ip dscp af41 class-map match-all TP match ip dscp cs4 class-map match-all signalling match ip dscp cs3 class-map match-any missioncritical-call-signaling match ip dscp cs3 af31 cs6 cs7 class-map match-any prec4 description critical business applications like sap match ip precedence 4 class-map match-any prec5 description real time interactive data like voice and video conferencing match ip precedence 5 class-map match-any prec1 description priority batch applications like notes match ip precedence 1 class-map match-any prec2 description interactive applications like intranet match ip precedence 2 class-map match-any prec3 description low volume interactive sessions match ip precedence 3 class-map match-any priority-telepresence match ip dscp cs4 class-map match-any realtime-voice match ip dscp cs5 ef class-map match-all voice match ip dscp ef ! ! policy-map TP class voice priority 15300 class class-default policy-map test class voip class voice priority 128 class TP bandwidth percent 78 class signalling bandwidth percent 5 class class-default fair-queue policy-map HQoS class class-default shape average 150000000 service-policy test policy-map etm-VMWare-TP class realtime-voice priority 128 police cir 128000 conform-action transmit exceed-action drop policy-map PIP description voice and TelePresence video



class prec5 priority percent 10 class telepresence bandwidth percent 40 class prec4 bandwidth percent 10 class prec3 bandwidth percent 5 class prec2 bandwidth percent 5 class prec1 bandwidth percent 5 class class-default ! ! ! ! ! interface GigabitEthernet0/0 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$ ip address 172.20.150.200 255.255.255.0 duplex auto speed auto media-type rj45 no mop enabled ! interface GigabitEthernet0/1 no ip address shutdown duplex full speed 100 media-type rj45 service-policy output HQoS ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! interface Serial0/1/1:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 172.20.150.1 ! ip http server ip http access-class 23 ip http authentication local no ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000



! ! ! ip prefix-list test seq 5 permit 10.1.1.1/32 access-list 23 permit 10.10.10.0 0.0.0.7 access-list 100 permit udp host 10.1.1.1 host 20.1.1.1 dscp af41 log ! ! ! ! control-plane ! ! ! voice-port 0/0/0 station-id name Chinh station-id number 4155265001 caller-id enable ! voice-port 0/0/1 station-id number 7750 caller-id enable ! voice-port 0/0/2 ! voice-port 0/0/3 ! voice-port 0/1/0:23 ! voice-port 0/1/1:23 ! voice-port 0/2/0 timing hookflash-out 50 ! voice-port 0/2/1 ! voice-port 0/2/2 ! voice-port 0/2/3 ! ccm-manager mgcp ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server CM-KINGS ccm-manager config ! mgcp call-agent CM-KINGS 2427 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp rtp unreachable timeout 1000 action notify mgcp modem passthrough voip mode nse mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package no mgcp package-capability res-package no mgcp package-capability fxr-package no mgcp timer receive-rtcp mgcp sdp simple



mgcp fax t38 ecm mgcp fax t38 inhibit mgcp rtp payload-type g726r16 static mgcp bind control source-interface GigabitEthernet0/0 mgcp bind media source-interface GigabitEthernet0/0 mgcp behavior g729-variants static-pt ! mgcp profile default ! sccp local GigabitEthernet0/0 sccp ccm 172.20.150.251 identifier 1 version 6.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register mtp0123456789ab ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP ! ! dial-peer voice 999001 pots service mgcpapp port 0/0/1 ! dial-peer voice 10 voip ! dial-peer voice 1 pots ! dial-peer voice 4100 pots description DP for call from OCS to Avaya destination-pattern 41.. incoming called-number 2113 direct-inward-dial port 0/1/0:23 forward-digits all ! dial-peer voice 2113 voip description DP for call from Avaya to OCS destination-pattern 2113 voice-class codec 1 session protocol sipv2 session target ipv4:172.20.150.251 session transport tcp incoming called-number 41.. dtmf-relay rtp-nte ! dial-peer voice 2 pots description DP for DTMF OCS call to FXS port destination-pattern 775. port 0/0/1 forward-digits 0



! ! sip-ua ! ! telephony-service max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! ephone-hunt 1 longest-idle ! ! ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 exec-timeout 0 0 privilege level 15 password cisco login local transport input telnet line vty 5 15 access-class 23 in exec-timeout 0 0 privilege level 15 password cisco login local transport input telnet ! scheduler allocate 20000 1000 end



AcronymsAcronymsAcronymsAcronyms

Cisco UCM Cisco Unified Communications Manager CFA Call Forwarding Always CFB Call Forwarding Busy CFNR Call Forwarding No Reply DN Directory Number DND Do-Not-Disturb DNS Domain Name System DTMF Dual Tone MultiFrequency DTP DeskTop Phone MOC Microsoft Office Communicator MCS Multimedia Communication Server MWI Message Waiting Indicator OCS Office Communications Server SCCP Skinny Protocol SIP Session Initiation Protocol TCP Transmission Control Protocol TLS Transport Layer Security XML Extensible Markup Language QoS Quality of Service GW Gateway S/W Software DB Database



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