SIP-Based Trunk Managed Voice Services Solution Design and Implementation Guide (Intenal Version)
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Table Of Contents
SIP-Based Trunk Managed Voice Services Solution Design and Implementation Guide
Cisco Unified Communications Manager
Voice Mail at the Enterprise Headquarter Site
Cisco Adaptive Security Appliance Firewall Appliance
Cisco Survivable Remote Site Telephony
SIP Trunking Design Considerations
SIP Delayed Offer and Early Offer
SIP Trunk Redundancy and Load Balancing
Encryption of Media and Signaling
Best Practices for SIP Trunk Implementation Using Cisco UBE
Obtaining Documentation and Submitting a Service Request
Overview of Test Configurations
Enterprise 1 and Branch 1 Components
Cisco Unified CM 6.1.0.9901-372 Caveats
Cisco UBE Version 1.2 (IOS Release 15.1(1)T) Caveats
Cisco Unity Express 3.2 Caveats
Cisco ASA 8.0(4) CaveatsHigh-Level Operation
HQ Call Flow to Enterprise Offsite Remote Endpoint
Branch 1 Call Flow to Enterprise Offsite Remote Endpoint
High-Level Configuration Summaries
Enterprise 1 HQ Cisco UBE Example Configuration
Enterprise 1 HQ Cisco Unified CM Example Configuration
Configuring the Cisco Unified CM System Parameters
System: Device Pool Parameters
Configuring the Cisco Unified CM Call Routing Parameters
Call Routing: Route/Hunt Parameters
Call Routing: Class of Control Parameters
Configuring the Cisco Unified CM Media Resources Parameters
Media Resources: Annunciator Parameters
Media Resources: Conference Bridge Parameters
Media Resources: Media Termination Point Parameters
Media Resources: Music on Hold Server Parameters
Media Resources: Transcoder Parameters
Media Resources: Media Resource Group Parameters
Media Resources: Media Resource Group List Parameters
Configuring the Cisco Unified CM Voice Mail Parameters
Voice Mail: Cisco Voice Mail Port Parameters
Voice Mail: Message Waiting Parameters
Voice Mail: Voice Mail Pilot Parameters
Voice Mail: Voice Mail Profile Parameters
Configuring the Cisco Unified CM Device Parameters
Enterprise 1 HQ Cisco Unity and Cisco Unity Express Example Configuration
Enterprise 1 HQ and Cisco VG224 Analog Phone Gateway Example Configuration
Enterprise 1 HQ Cisco ASA Firewall Example Configuration
Branch 1 Cisco UBE, TDM Gateway, and Cisco Unified SRST Example Configuration
Branch 1 Cisco Unity Express 3.2 and Cisco Unified CM Example Configuration
Cisco Unified Border Element Performance Summary
SIP-Based Trunk Managed Voice Services Solution Design and Implementation Guide
First Published: April 21, 2010, OL-18623-02
Last Updated: April 21, 2010Contents
Cisco Unified Communications Manager
Voice Mail at the Enterprise Headquarter Site
Cisco Adaptive Security Appliance Firewall Appliance
Cisco Survivable Remote Site Telephony
SIP Trunking Design Considerations
Encryption of Media and Signaling
Best Practices for SIP Trunk Implementation Using Cisco UBE
SIP Delayed Offer and Early Offer
SIP Trunk Redundancy and Load Balancing
Obtaining Documentation and Submitting a Service Request
Overview of Test Configurations
Cisco ASA 8.0(4) CaveatsHigh-Level Operation
Enterprise 1 HQ Cisco UBE Example Configuration
Enterprise 1 HQ Cisco Unified CM Example Configuration
Enterprise 1 HQ Cisco Unity and Cisco Unity Express Example Configuration
Enterprise 1 HQ and Cisco VG224 Analog Phone Gateway Example Configuration
Enterprise 1 HQ Cisco ASA Firewall Example Configuration
Branch 1 Cisco UBE, TDM Gateway, and Cisco Unified SRST Example Configuration
Branch 1 Cisco Unity Express 3.2 and Cisco Unified CM Example Configuration
Introduction
Cisco Unified Communications delivers fully integrated communications systems by enabling data and voice to be transmitted over a single network infrastructure using standards-based Internet Protocol (IP). Leveraging the framework provided by Cisco IP hardware and software products, Cisco Unified Communications delivers unparalleled performance and capabilities to address current and emerging communications needs in service provider, enterprise, and commercial business environments.
This guide discusses a solution network design to enable enterprise Session Initiation Protocol (SIP) trunk deployment with Cisco Unified Communications Manager (Cisco Unified CM) and Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST), one of the several SIP trunk solutions that Cisco is developing. The model of enterprise SIP trunk development with Cisco Unified CM and Cisco Unified SRST is especially geared for large enterprises with many branch offices. In this distributed model, the service provider (SP) furnishes the SIP trunk services for the enterprise to connect the enterprise headquarter with its enterprise branch offices. At the enterprise headquarter, Cisco Unified CM provides call control for voice services. Remote enterprise branch offices have Cisco Unified SRST deployed for voice services. The Cisco Integrated Services Router (Cisco ISR) running the Cisco Unified Border Element (Cisco UBE) is placed at the edge of the network. Cisco UBE plays an important role in serving multiple functions when connecting to other networks.
This design guide discusses the components deployed in the network, and provides sample router configurations for the Cisco UBE functions tested for the features included in this document. This guide is an update to the existing SRND and validates the Cisco UBE functions on the second generation Cisco Integrated Services Router (Cisco ISR-G2) 29xx/39xx and 3945E platforms, All other solution components remain unchanged.
Use this information to deploy enterprise SIP trunks with Cisco Unified CM and Cisco Unified SRST using service provider networks.
Network Topology
The components of the enterprise SIP trunk deployment with Cisco Unified CM and Cisco Unified SRST network topology is show in Figure 1. The service provider components are listed for completeness only and are not included in this guide.
Enterprise Headquarter
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Enterprise 1 HQ Cisco UBE Example Configuration
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Enterprise Branch
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Service Provider
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Figure 1 Enterprise SIP Trunk Deployments Cisco Unified CM and Cisco Unified SRST with Cisco UBE
Prerequisites
Prerequisites are grouped into the following sections:
Components Used
The information in this guide is based on the software and hardware versions listed in the following sections. The configuration shown in this guide was created through the use of the devices in a specific lab environment. This section includes prerequisites for the following components:
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Cisco Unified Communications Manager
The Cisco Unified CM at the enterprise headquarter site provides call control to voice services at the headquarter site and the branch offices. The Cisco Unified CM was tested using version 6.1.3.
Cisco Unified Border Element
A Cisco 3945 and 3945A series platforms were tested with Cisco IOS Release 15.1(1)T and Cisco UBE version 1.4. The Cisco 2900 series Integrated Services Router (Cisco ISR) can also be used as a Cisco UBE.
SCCP Analog Voice Gateway
A Cisco VG224 analog voice gateway was used at the enterprise headquarter site to provide connectivity to analog phones and fax machines. The Cisco VG224 analog voice gateway was tested with Cisco IOS Release 15.1(1)T.
Voice Mail at the Enterprise Headquarter Site
Voice mail at the enterprise headquarter site is provided by the Cisco Unity voice mail server, which was tested with version 3.2.
Cisco Adaptive Security Appliance Firewall Appliance
A Cisco ASA firewall appliance was placed at the ingress from the service provider servicing the enterprise headquarter site. It was tested with Cisco ASA 8.0(4).
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Cisco Survivable Remote Site Telephony
A Cisco Unified SRST router is placed at the enterprise branch site. In addition to the Cisco Unified SRST functions, this router provides Cisco UBE, Cisco IOS firewall, conferencing transcoding, MTP, voice mail using Cisco Unity Express, TDM, and gateway functions. A Cisco 3800 series platform was tested with Cisco IOS Release 15.1(1)T. Cisco Unity Express was tested with version 3.2. The Cisco 2800 series Integrated Services Router (Cisco ISR) can also be used as an Cisco Unified SRST router.
SIP Call Agent
Various SIP call agents can be used for the feature functionality discussed in this guide. For testing purposes, a BroadSoft call agent release 14 SP7 was used.
The BroadSoft call agent uses the BroadWorks platform. The typical deployment is comprised of three servers installed in a clustered or redundant format:
PSTN Hop-Off Gateway
A Cisco AS5000 series gateway with PRI trunks was used and tested with Cisco IOS Release 15.1(1)T. You can use other software releases later than Cisco IOS Release 15.1(1)T or other gateway platforms.
Cisco IOS Software Releases
The test results described in this guide for the Cisco Unified Border Element were conducted using Cisco IOS Release15.1(1)T. We recommend Cisco IOS Release 15.1(1)T or later releases for the deployment of the features described in this guide.
Conventions
Refer to Cisco Technical Tips Conventions for information on document conventions.
Solution Description
The enterprise SIP trunk deployment with the Cisco Unified CM and Cisco Unified SRST solution topology allows the enterprise headquarter site to provide voice services from Cisco Unified CM to remote enterprise branch offices using SIP trunks from service providers. The enterprise branch offices are equipped with Cisco Unified SRST routers.
When Cisco Unified CM fails, but the WAN connection remains active and SRST takes over, the remote phones are able to make WAN calls through SIP to the call agent. If a WAN connectivity failure occurs, the enterprise branch offices can continue to maintain basic IP phone and PSTN services.
The focus of services using this solution are:
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The following topics describe the solution:
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Feature Summary
The features listed in this section were tested as part of the solution configuration.
Enterprise Headquarter Site Features
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Enterprise Branch Offices Features
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Service Provider Features
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Basic Phone Features Served in the Topology
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SIP Trunking Design Considerations
SIP trunking design considerations described in the following sections should be assessed when deploying SIP trunks.
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DTMF Transport
There are several ways of transporting DTMF information between SIP endpoints. In general, these methods can be classified as Out of Band (OOB) and In Band (IB) signaling. IB DTMF transport methods send either raw or signaled DTMF tones within the RTP stream and need to be processed by the endpoints that generate or receive them.
OOB signaling methods transport DTMF tones outside of the RTP steam, either directly to and from the endpoints or using a Call Agent, such as the Communications Manager, which interprets and forwards these tones as required.
OOB SIP DTMF signaling methods include:
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KPML (RFC 4730) is the preferred OOB signaling method used by Cisco. KPML is supported on Advanced Cisco 79X1 Series IP Phones, Cisco Unified CM, and Cisco IOS Gateways (Cisco IOS Release 15.1 and later).
Unsolicited Notify is a proprietary DTMF transport method used only on Cisco IOS Gateways (Cisco IOS Release 12.2 and later).
IB DTMF transport methods send DTMF tones as either raw tones in the RTP media stream or as signaled tones in the RTP payload, using RFC 2833.
With SIP product vendors, RFC 2833 has become the predominant method of sending and receiving DTMF tones and is supported by the majority of Cisco voice products.
Because IB signaling methods send DTMF tones in the RTP media stream, the SIP endpoints in a session must either support the transport method used (for example, RFC 2833) or provide a method of intercepting this in band signaling and converting it. That is, if two endpoints are using a B2BUA as the call control agent (such as the Communications Manager) and they negotiate different DTMF transport methods, then the call control agent determines how these DTMF transport differences are handled. With Communications Manager, a DTMF transport mismatch (for example, IB to OOB DTMF is resolved by inserting a transcoder).
SIP Delayed Offer and Early Offer
RFC 3261 defines two ways that Session Description Protocol (SDP) messages can be sent in the offer and answer, commonly known as Delayed Offer and Early Offer, which are mandatory requirements in the specification. In the simplest terms, an initial SIP Invite sent with SDP in the message body defines an Early Offer; whereas, an initial SIP Invite sent without SDP in the message body defines a Delayed Offer. In an Early Offer, the session initiator sends its capabilities in the SDP contained in the initial invite (for example, codecs supported). In a Delayed Offer, the session initiator does not send its capabilities in the initial invite and waits for the called device to send its capabilities first.
Cisco UBE uses the SIP Offer/Answer model for establishing SIP sessions, as defined in RFC 3264. In this context, an Offer is contained in the SDP fields sent in the body of a SIP message.
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Early Media Cut Through
The terms Early Offer and Early Media are often confused.
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In certain circumstances, a SIP session can require that a media path be set up prior to completing a connection. To this end, the SIP protocol allows the establishment of Early Media after the initial Offer has been received by an endpoint. The reasons for using Early Media vary.
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Both Early Offer and Delayed Offer calls support Early Media. Early Offer calls can typically stream Early Media after exchanging two messages (Invite with SDP and Trying). Delayed Offer calls can typically stream Early Media after exchanging four messages (Invite without SDP, 100 Trying, Session Progress with SDP and PRACK).
If Cisco UBE is configured to do DO->EO conversion, ensure that PRACK is enabled on CUCM, for call flows involving early media cut-through (18x w/SDP) to work seamless.
SIP Trunk Transport Protocols
SIP Trunks can use either TCP or UDP as a message transport protocol. As a reliable, connection orientated protocol that maintains the connection state per SIP dialogue, TCP is preferred. However, TCP has a higher segment overhead, uses more bandwidth than UDP, and has a higher packet overhead. These TCP overhead features increase call setup times when compared with UDP, which is connectionless and relies on the SIP stack to maintain its state and reliability.
If your network is prone to packet loss, use TCP. If the networks do not experience packet loss, use UDP.
Monitoring SIP Trunk State
SIP servers can monitor individual SIP dialogues either by using the dialogue TCP connection or within the SIP stack itself (for example, for UDP based transport). In a Cisco Unified CM environment, use this per-call trunk state tracking feature in conjunction with Cisco Unified CM Route Groups and Route Lists to route calls over multiple SIP trunks. Trunk state is monitored and state changes are detected on a per-call basis. Successive trunk connections are attempted when the first trunk and subsequently selected trunks are down.
To overcome the limitations of per-call, per trunk state detection, the following methods can be used to monitor the state and detect the state changes of each end of a SIP trunk:
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Cisco UBE sends an Out of Dialogue OPTIONS message to the device at the far-end of the SIP trunk to determine its state. The OPTIONS method is used as an application-level ping. The returned ping response is generally not as important as the fact that the trunk has confirmed that it is alive. Cisco Unified CM SIP trunks support the receipt of OPTIONS messages but do not send OPTIONS messages as keepalives. Cisco Unified CM version 5.x SIP trunks respond to OPTIONS messages with a "405—Method Not Acceptable" response. In Cisco Unified CM version 6.0.1, SIP trunks respond to an OPTIONS message with a "200—OK" response.
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SIP Trunk Redundancy and Load Balancing
Redundancy can be achieved by combining the call admission control (CAC) features of IOS. In general, CAC can be applied based on IP address reachability, Total Memory, Total Calls, Total CPU, IP circuit max-calls, and max-connections. The following show several methods used to achieve redundancy based on:
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Dial-peer preferences and Dial-peer Hunting
Use the following CLI example to achieve redundancy based on dial-peer preferences and dial-peer hunting:
!dial-peer voice 3670000 voipdescription "first hunting for 3670000 to ent2-hq-ipip"destination-pattern 240367....session protocol sipv2session target ipv4:10.10.11.36codec g711ulaw!dial-peer voice 36700 voipdescription "second hunting for 3670000 to ent2-hq-ipip"destination-pattern 240367....preference 1session protocol sipv2session target ipv4:10.10.11.37codec g711ulaw!DNS SRV
Use the setup example shown in Figure 2 into achieve redundancy based on DNS SRV.
Figure 2 SIP Network Redundancy and Scaling Based on DNS SRV
Route List & Route Group option from CCM
To achieve redundancy based on route list and route group using Cisco Unified CM, complete the following steps:
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Figure 3 Configuring Route Groups
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Figure 4 Configuring A Route List for Route Groups
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Figure 5 Configuring A Route List Under Route Pattern Gateway or Route List
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Figure 6 Configuring Max-Con
IP Connectivity
The SIP trunks are typically provided by service providers (SPs). SP voice services are offered using a SIP trunk that uses the same physical IP interface also used to deliver data services. The options for the physical connection of SIP trunks from the SPs are shown in Table 1.
The sample configuration in the "Configurations" section shows a Gigabit Ethernet interface.
Some service providers that offer both data and voice services over a single IP interface also offer MPLS services. With MPLS services, voice packets must be sent with an MPLS label so that the service provider can terminate the traffic, and data marked with a different label can be tunneled through the backbone network. Marking voice traffic with an MPLS label requires the Virtual Routing and Forwarding (VRF)-Aware voice feature available on the Cisco ISRs in Cisco IOS Release 15.1(1)T.
Quality of Service
Quality of Service (QoS) is a fundamental requirement for any IP interface that carries voice traffic. Several specific QoS considerations and their configurations are discussed in this section:
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Congestion Management
When you use a single connection for both voice and data, you must carefully consider congestion management and bandwidth allocation to prevent data flows from affecting voice quality.
VoIP signaling and media traffic can be identified and classified as priority traffic using the QoS tools available within Cisco IOS software. Use Low Latency Queuing (LLQ) for media traffic streams. During congestion, LLQ queues restrict throughput to the configured bandwidth and packets exceeding this bandwidth are dropped. Therefore, signaling traffic should use class-based weighted fair queuing (CBWFQ), because signaling traffic bursts during call setup and teardown. The configurations for LLQ and CBWFQ are shown in the "Configurations" section. See Quality of Service for Voice Over IP for more information.
You can estimate the bandwidth to allocate to voice traffic by considering:
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The service provider can limit the maximum number of calls allowed across the SIP trunk based on the CAC techniques discussed in the "Billing and Management" section. This maximum number of calls allowed can be part of the service level agreement (SLA) between the service provider and the end customer.
When a Layer 2 connection technology, like Frame Relay or ATM, is used, additional traffic shaping and traffic management mechanisms must be deployed to ensure QoS on the egress interface. See Configuring Frame Relay for more information.
Packet Marking
You must set appropriate differentiated services code point (DSCP) values on the media and signaling packets leaving the SIP trunk from the customer premises to receive the desired service level in the service provider network. By default, Cisco IOS software on the CPE router marks voice media packets, sourced on the router, with DSCP EF (101110) for expedited forwarding and signaling packets, sourced on the router, with DSCP AF31 (011010) for assured forwarding.
QoS policies may use either DSCP or IP precedence to classify voice packets. IP precedence interprets the low order three bits of the 6-bit DSCP value. In this way DSCP EF maps to CS5, while DSCP AD31 maps to CS3, which are appropriate IP precedence settings for voice media and signaling traffic.
Call Admission Control
Different types of Call Admission Control (CAC) are used in this solution. CAC can be based on bandwidth, maximum connections, CPU load, or memory available. CAC can be enabled at Cisco Unified CM or Cisco UBE.
Bandwidth-based CAC monitors the amount of bandwidth available in the network and controls routing of calls accordingly. This provides guaranteed control of bandwidth usage for voice calls. On Cisco Unified CM, bandwidth-based CAC is available and tested.
The number of simultaneous outbound calls can also be limited by the max-conn command on the VoIP dial peer used to route calls from the Cisco UBE router to the service provider network. This is the mechanism tested in the configuration example given in this guide.
The Cisco UBE can control the number of calls by setting the CPU load or memory available. This is configurable on the Cisco UBE by setting the threshold such that CAC is triggered when the threshold is reached.
The service provider can also control the total number of inbound and outbound calls from the SIP feature server, which is the best place for CAC policies to be implemented.
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Delay
The telephone industry standard ITU-T G.114 recommends the maximum desired one-way delay for a voice packet be no more than 150 milliseconds (ms). With a round-trip delay of 300 ms or more, users can experience annoying talk-over. In addition to congestion management with proper queuing techniques, you can use link fragmentation and interleaving (LFI) on slower access links to ensure that the end-to-end delay budget for voice packets is met. LFI is usually necessary on links of less than 768K access speeds.
Variable delay in packet rate results in jitter. The jitter buffer in Cisco voice gateways runs in an adaptive mode and can remove the jitter from the packet flow for moderate end-to-end jitter in the network. See Understanding Jitter in Packet Voice Networks (Cisco IOS Platforms) for more information on jitter. Delay can also cause echo.
Echo
Echo is caused by a time-division multiplexing (TDM) connection, or acoustic echo resulting from IP connections and endpoints. An improperly insulated phone, headset, or speakerphone could be the cause of echo experienced across a SIP trunk call. The analog phone user can also hear echo because of a very hot, or very high volume, signal on the TDM interface. Echo Analysis for Voice over IP explains how to adjust the settings for the voice port to eliminate echo caused by a hot signal and contains details on troubleshooting the source of echo. Delayed echo could be from the PSTN connectivity in the service provider's network. Cancel this echo on the PSTN gateway.
Voice Mail
Voice mail is provided by the Cisco Unity server at the enterprise headquarter site. At the enterprise branch offices, voice mail is provided by Cisco Unity Express embedded in the Cisco Unified SRST router.
The service provider can offer voice mail services using a hosted server. In this configuration, the service provider SIP server is responsible for functions such as call forward busy, call forward no answer, and Message Waiting Indicator (MWI).
Dial Plan
In this solution topology, the voice services are provided by the service provider using a call agent. The dial plan is also controlled by the service provider. The configuration shows the call routing configuration for VoIP dial peers needed on the Cisco UBE.
Security
The following security features are included in the solution network design:
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Authentication
SIP registration and call method authentication can be provided using Digest Authentication. This method uses a single username and password for the entire SIP trunk, as shown in the "Configurations" section. The password is encrypted using Message Digest 5 (MD5).
Encryption of Media and Signaling
Virtual Private Network (VPN) technology can be used to encrypt the media and signaling streams between the Cisco UBE router and the core network. Cisco UBE also supports Transport Layer Security (TLS) and Secure RTP (SRTP) internally between phones and the router.
Firewall
At the enterprise headquarter site, either the Cisco ASA firewall appliance or Cisco IOS Zone-based firewall, can be used to defend against outside attacks from the IP interface entering the headquarter. At the enterprise branch offices, the Cisco IOS Zone-based firewall features in the Cisco Unified SRST router are used. The firewall serves as a checkpoint for the customer LAN traffic exiting from the router to the service provider network.
Access control lists (ACLs) are required to filter out unwanted traffic on physical links to the Internet. These ACLs are used primarily to stop unauthorized access, Denial of Service (DoS) attacks, or distributed DoS (DDoS) attacks that originate from the service provider or a network connected to the service provider, and also to prevent intrusions and data theft.
In this test configuration, the Cisco ASA firewall appliance was used at the enterprise headquarter site, and Cisco IOS firewall features were used at the enterprise branch offices.
Failover and Redundancy
If a complete SIP trunk failure or IP interface failure occurs, backup PSTN lines connected directly to Cisco Unified SRST can be used for PSTN access. In the Cisco Unified SRST router configuration shown in the "Configurations" section, backup PSTN access was tested for alternate call routing when SIP trunk access was down.
At the enterprise headquarter site, PSTN access was not explicitly tested on SIP trunk failure in this test configuration. PSTN access at the enterprise headquarter site can be deployed in a similar way because it was tested in other test configurations on SIP trunk failure.
Fax and Modem
Fax pass-through and modem pass-through calls were tested between the enterprise headquarter site and branch offices and to the PSTN hop-off gateway. Fax and modem calls were tested with the G.711 codec.
Billing and Management
Typically, the service provider is able to do billing without using any information from the managed Cisco UBE router.
Each call through the Cisco UBE router is considered to have two call legs. The start and stop records are generated for each call leg and can be polled through Simple Network Management Protocol (SNMP) using the DIAL-CONTROL-MIB. For more information, see the following documents:
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Best Practices for SIP Trunk Implementation Using Cisco UBE
By using the following Cisco UBE configuration methods, you can achieve a more effective SIP trunk topology implementation.
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description *** incoming calls from Service Provider ***
incoming called number xxx
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description *** outgoing calls to Enterprise ***
destination-pattern xxx
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To ensure minimum Post Dial Delay during failover situations, fine tune the sip-ua retry xxx parameters, where xxx is the request name and response code. We recommend the value for INVITEs as retry invite 2.
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The Layer 3 and Layer 7 embedded SIP addresses can be unpredictable when Cisco HSRP is enabled. Refer to the caveats section for exact Bug-ID.
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SIP Profiles is a very powerful SIP message normalization and protocol repair tool that can quickly fix or create a workaround to minor interoperability issues when two SIP implementations communicate with each other. This feature is available in Cisco IOS 12.4(15)XZ and Cisco IOS 12.4(20)T and later.
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Adjust the payload types for DTMF as needed when the default Cisco values are in conflict (for example, PT 96 is used for RFC 2833, which is by default reserved for cisco fax-relay).
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You can configure CLI commands such as allow term.
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Caveats
In general, the following global caveats exist with this solution:
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Configurations
The "Appendix: Enterprise 1 and Branch 1 SIP-Based Trunk Managed Voice Services Solution Example Configurations" section provides configuration examples, screen figures, and other helpful information you need to configure the features on the Cisco UBE router at the edge of the service provider network described in this guide.
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Configuration Verification
Use the following show commands to display and verify your Cisco UBE configuration:
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The firewall configuration can be verified with the following commands:
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Troubleshooting
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Use the following debug commands to troubleshoot your configuration:
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This command shows all SIP Service Provider Interface (SPI) message tracing. It traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.
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This command enables all SIP-related debugging including:
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This command displays all application debug messages, including Application Framework (AFW) and DSAPP debugs.
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This command traces the execution path through the call control API, which serves as the interface between the call session application and the underlying network-specific software. You can use the output from this command to understand how calls are being handled by the voice gateway.
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This command enables Media Termination Point (MTP) debugging.
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This command displays debugging information for SCCP events and its related applications transcoding and conferencing.
Related Information
The following information is referenced in this guide:
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Obtaining Documentation and Submitting a Service Request
For information on obtaining documentation, submitting a service request, and gathering additional information, see the monthly What's New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at:
http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html
Subscribe to the What's New in Cisco Product Documentation as a Really Simple Syndication (RSS) feed and set content to be delivered directly to your desktop using a reader application. The RSS feeds are a free service and Cisco currently supports RSS Version 2.0.
Appendix: Enterprise 1 and Branch 1 SIP-Based Trunk Managed Voice Services Solution Example Configurations
This appendix contains configuration examples to configure a SIP-based managed voice services solution using the Cisco Unified Border Element, Cisco Unified Communications Manager, Cisco Unity, and Cisco Unity Express, depending on your configuration requirements.
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Overview of Test Configurations
The following main components are used in the Voice Enterprise 1 configuration:
Enterprise 1 HQ Components
The main components of the Enterprise 1 Headquarters (HQ) include:
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Enterprise 1 and Branch 1 Components
The main components of the Enterprise 1 and Branch 1 include:
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Caveats
The following caveats apply to the SIP-based Trunk Voice Enterprise 1solution:
Global Caveats
In general, the following global caveats exist with this solution:
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Cisco Unified CM 6.1.0.9901-372 Caveats
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Cisco UBE Version 1.2 (IOS Release 15.1(1)T) Caveats
Cisco Unity 5.0(1) Caveats
To view the caveats for Cisco Unity 5.0(1), see Release Notes for Cisco Unity Release 5.0(1).
Cisco Unity Express 3.2 Caveats
To view the caveats for Cisco Unity Express 3.2, see Release Notes for Cisco Unity Express 3.2.
Cisco ASA 8.0(4) CaveatsHigh-Level Operation
Users trying to configure the Voice Enterprise 1 topology should be familiar with networking in general and the specific configurations of the following Cisco applications:
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CAll Flow Within Enterprise 1
All endpoints (Cisco Unified CM, HQ/Branch Cisco UBEs, IP phones, and so on) in the Voice Enterprise 1 network are configured to be routable. Calls within the enterprise use SCCP/MGCP for call control.
During normal operation, call flow from HQ to Branch 1 are as follows:
IP/VG224 FXS Phone (over SCCP) > Cisco Unified CM (over SCCP/MGCP) > IP/Branch Cisco UBE FXS Phone
During normal operation, Branch l call flows to HQ is in the reverse direction.
HQ Call Flow to Enterprise Offsite Remote Endpoint
During normal operation, call flow from HQ to outside of the enterprise is as follows:
IP/VG224 FXS phone (over SCCP) > Cisco Unified CM (over SIP) > HQ Cisco UBE (over SIP) > Service Provider SIP Proxy Server
During normal operation, external call flow to the enterprise HQ is in the reverse direction.
Branch 1 Call Flow to Enterprise Offsite Remote Endpoint
Call flow from Branch 1 to outside of the enterprise would be as follows:
IP/Branch Cisco UBE FXS phone (over SCCP/MGCP) > Cisco Unified CM (over SIP) > Branch Cisco UBE (over SIP) > Service Provider SIP Proxy Server
For normal operation, external call flow to the enterprise Branch 1 is in the reverse direction.
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Cisco Unified CM is used to control the number of uplink calls (CAC—bandwidth) for both the enterprise HQ and branch.
For purposes of security, the Cisco ASA can be placed at the front end of the HQ Cisco UBE.
High-Level Configuration Summaries
The following topics summarize the scope of a current enterprise solution:
Protocols
The following is a list of protocols used between components:
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Codecs
The following is a list of codecs used between components:
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DSP Farms
Separate DSP farms are installed and configured on the enterprise HQ and Branch Cisco UBEs. Although only conference resources are used for these solutions, MTP and Transcoder resources are also configured and are registered to Cisco Unified CM for example purposes only.
Supplementary Services
The following is a list of supplementary services:
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Call Admission Control
The call admission control (CAC) restrictions that are imposed by Cisco Unified CM for the whole enterprise are as follows:
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Test Topology
Figure 7 shows the setup test topology used in example configurations described in the following sections.
Figure 7 Test Topology
Example Configuration Details
The IP addresses used with SIP in the network are as follows:
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The selection of the static codec for either a voice or fax call is implemented by tightly integrating the configurations of Cisco Unified CM and site Cisco UBE. For the DO-to-EO to originate from the originator's local Cisco UBE and for the correct codec to be used with the Service Provider SIP proxy server, the following configuration example has been set up:
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Calls terminating at the enterprise are also tightly controlled as to whether they are IP phone (g729r8) or FXS phone (g711u), where the latter is mainly used for fax/modem purposes. Received calls that do not match these criteria are rejected.
The dial-plan for the enterprise HQ and the Branch sites can be any global numbering plan. In the following example, the same area code was used for the enterprise HQ 1 and the Branch 1.
Enterprise 1 HQ Cisco UBE Example Configuration
The following example shows a command-line interface (CLI) configuration example for the enterprise 1 HQ Cisco Unified Border Element for the test topology described in Figure 7.
Ent1_HQ_CUBE1#!voice-card 0dspfarmdsp services dspfarm!voice service voipmode border-elementallow-connections h323 to h323allow-connections h323 to sipallow-connections sip to h323allow-connections sip to sipsignaling forward unconditionalfax protocol pass-through g711ulawmodem passthrough nse codec g711ulawh323emptycapabilityh245 passthru tcsnonstd-passthrusipbind control source-interface Loopback0bind media source-interface Loopback0min-se 2000header-passing error-passthruoptions-ping 1200listen-port non-secure 5090midcall-signaling passthru!voice translation-rule 1rule 1 /^61/ /1/rule 2 /^71/ /1/!voice translation-profile OUTGOING-SIP-TRK-DIGIT-STRIPtranslate called 1!!interface Loopback0ip address 10.10.11.151 255.255.255.255!interface GigabitEthernet0/0ip address 10.40.97.1 255.255.255.0duplex fullspeed 100media-type rj45no keepalive!interface GigabitEthernet0/1ip address 10.40.99.2 255.255.255.0duplex fullspeed 100media-type rj45no keepalive!ip rtcp report interval 9000!sccp local GigabitEthernet0/0sccp ccm 10.40.97.2 identifier 5 priority 1 version 6.0sccp!sccp ccm group 10associate ccm 5 priority 1associate profile 10 register MTP111222333associate profile 12 register CON111222333associate profile 11 register XCODE111222333!dspfarm profile 11 transcodecodec g711ulawcodec g729r8maximum sessions 10associate application SCCP!dspfarm profile 12 conferencedescription conference bridgecodec g711ulawcodec g729r8maximum sessions 10associate application SCCP!dspfarm profile 10 mtpcodec g711ulawmaximum sessions software 5associate application SCCP!dial-peer voice 2000 voipdescription *** Voice: LAN to WAN - Incoming Dial-Peer ***huntstopcodec g729r8session protocol sipv2incoming called-number 6Tdtmf-relay rtp-nte digit-dropno vad!dial-peer voice 2001 voipdescription *** Voice: LAN to WAN - Outgoing Dial-Peer ***translation-profile outgoing OUTGOING-SIP-TRK-DIGIT-STRIPhuntstopdestination-pattern 6Tcodec g729r8voice-class sip early-offer forcedmax-redirects 5session protocol sipv2session target ipv4:10.3.33.22dtmf-relay rtp-nte digit-dropno vad!dial-peer voice 2100 voipdescription *** Voice: WAN to LAN - Incoming Dial-Peer ***huntstopcodec g729r8session protocol sipv2incoming called-number 415Tdtmf-relay rtp-nte digit-dropno vad!dial-peer voice 2101 voipdescription *** Voice: WAN to LAN - Outgoing Dial-Peer ***huntstopdestination-pattern 415Tcodec g729r8max-redirects 5session protocol sipv2session target ipv4:10.40.97.2dtmf-relay rtp-nte digit-dropno vad!dial-peer voice 3000 voipdescription *** Fax: LAN to WAN - Incoming Dial-Peer ***huntstopsession protocol sipv2incoming called-number 7Tdtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 3001 voipdescription *** Fax: LAN to WAN - Outgoing Dial-Peer ***translation-profile outgoing OUTGOING-SIP-TRK-DIGIT-STRIPhuntstopdestination-pattern 7Tvoice-class sip early-offer forcedmax-redirects 5session protocol sipv2session target ipv4:10.3.33.22dtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 3100 voipdescription *** Fax: WAN to LAN - Incoming Dial-Peer ***huntstopsession protocol sipv2incoming called-number 415555105[0,1]dtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 3101 voipdescription *** Fax: WAN to LAN - Outgoing Dial-Peer ***huntstopdestination-pattern 415555105[0,1]max-redirects 5session protocol sipv2session target ipv4:10.40.97.2dtmf-relay rtp-nte digit-dropcodec g711ulawno vad!gatewaymedia-inactivity-criteria alltimer receive-rtcp 5timer receive-rtp 180!sip-uakeepalive target ipv4:10.3.33.22authentication username yyyy password 7 xxxxxxxxxxno remote-party-idretry invite 2retry bye 2retry cancel 2timers keepalive active 600reason-header overrideg729-annexb override!Ent1_HQ_CUBE1#Enterprise 1 HQ Cisco Unified CM Example Configuration
The following example shows the required field and parameter entries for example configuration of the Cisco Unified CM for the topology shown in Figure 7. Parameters are entered using the Cisco Unified CM GUI. The example parameters windows entries are shown in following sections:
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Configuring the Cisco Unified CM System Parameters
Use the Cisco Unified Communications Manager Administration window to configure system parameters. The system parameter example configurations are shown in the following sections:
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System: Server Parameters
To configure the system server parameters for the Cisco Unified CM, click System > Server in the Cisco Unified CM Administration window.
Figure 8 System Server Enterprise 1 HQ Cisco Unified CM Administration Window
System: Region Parameters
To configure the system region parameters for the Cisco Unified CM, click System > Region in the Cisco Unified CM Administration window.
Figure 9 System Region Cisco Unified CM Administration Window
Figure 10 System Region Default Cisco Unified CM Administration Window
Figure 11 System Region-Region Branch 1 Phones Analog Cisco Unified CM Administration Window
Figure 12 System Region-Region Branch 1 DSP Farm Cisco Unified CM Administration Window
Figure 13 System Region-Region Branch 1 DSP Farm Conference Cisco Unified CM Administration Window
Figure 14 System Region-Region Branch 1 DSP Farm Transcoder Cisco Unified CM Administration Window
Figure 15 System Region-Region Branch 1 Phones IP Cisco Unified CM Administration Window
Figure 16 System Region-Region HQ DSP Farm Cisco Unified CM Administration Window
Figure 17 System Region-Region HQ DSP Farm Conference Cisco Unified CM Administration Window
Figure 18 System Region-Region HQ DSP Farm Transcoder Cisco Unified CM Administration Window
Figure 19 System Region-Region HQ Phones Analog Cisco Unified CM Administration Window
Figure 20 System Region-Region HQ Phones IP Cisco Unified CM Administration Window
Figure 21 System Region-Region WAN Cisco Unified CM Administration Window
System: Device Pool Parameters
To configure the system device pool parameters for the Cisco Unified CM, click System > Device Pool in the Cisco Unified CM Administration window.
Figure 22 System Device Pool Cisco Unified CM Administration Window
Figure 23 System Device Pool Default Cisco Unified CM Administration Window
Figure 24 System Device Pool-DevicePool Branch 1 Analog Phones Cisco Unified CM Administration Window
Figure 25 System Device Pool-DevicePool Branch 1 DSP Farm Cisco Unified CM Administration Window
Figure 26 System Device Pool-DevicePool Branch 1 IP Phones Cisco Unified CM Administration Window
Figure 27 System Device Pool-DevicePool HQ Analog Phones Cisco Unified CM Administration Window
Figure 28 System Device Pool-DevicePool HQ DSP Farm Cisco Unified CM Administration Window
Figure 29 System Device Pool-DevicePool HQ IP Phones Cisco Unified CM Administration Window
Figure 30 System DevicePool-DevicePool WAN Cisco Unified CM Administration Window
System: Location Parameters
To configure the system location parameters for the Cisco Unified CM, click System > Location in the Cisco Unified CM Administration window.
Figure 31 System Location Cisco Unified CM Administration Window
Figure 32 System Location Hub Branch 1 Cisco Unified CM Administration Window
Figure 33 System Location Hub HQ Cisco Unified CM Administration Window
Figure 34 System Location Hub None Cisco Unified CM Administration Window
Figure 35 System Location-Location Trunk Branch 1 Cisco Unified CM Administration Window
Figure 36 System Location-Location Trunk HQ Cisco Unified CM Administration Window
System: SRST Parameters
To configure the system SRST parameters for the Cisco Unified CM, click System > SRST in the Cisco Unified CM Administration window.
Figure 37 System SRST-SRST Enterprise 1 Branch 1 Cisco Unified CM Administration Window
Configuring the Cisco Unified CM Call Routing Parameters
Use the Cisco Unified Communications Manager Administration window to configure call routing parameters. Call routing parameter example configurations are shown in the following sections:
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Call Routing: Route/Hunt Parameters
To configure call routing route/hunt parameters for the Cisco Unified CM, click Call Routing > Route/Hunt in the Cisco Unified CM Administration window.
Figure 38 Call Routing RouteHunt Route Pattern Cisco Unified CM Administration Window
Figure 39 Call Routing RouteHunt Route Pattern RP Ent 1 HQ IP Phone LongDistance Cisco Unified CM Admin Window
Figure 40 Call Routing RouteHunt Route Pattern RP Ent1 HQ Analog Phone LongDistance Administration Window
Figure 41 Call Routing RouteHunt Route Pattern RP Ent1 Br1 Analog Phone LongDistance Administration Window
Figure 42 Call Routing RouteHunt Route Pattern RP Ent1 Br1 IP Phone LongDistance Administration Window
Call Routing: Class of Control Parameters
To configure the call routing class of control parameters for the Cisco Unified CM, click
Call Routing > Class of Control in the Cisco Unified CM Administration window.Figure 43 Call Routing Class of Control Partition Cisco Unified CM Administration Window
Figure 44 Call Routing Class of Control Partition-Partition Br1 Phones Analog Administration Window
Figure 45 Call Routing Class of Control Partition-Partition Br1 Phones IP Cisco Unified CM Administration Window
Figure 46 Call Routing Class of Control Partition-Partition HQ Phones Analog Administration Window
Figure 47 Call Routing Class of Control Partition-Partition HQ Phones IP Cisco Unified CM Administration Window
Figure 48 Call Routing Class of Control CSS Cisco Unified CM Administration Window
Figure 49 Call Routing Class of Control CSS-CSS Branch 1 Phones Analog Cisco Unified CM Administration Window
Figure 50 Call Routing Class of Control CSS-CSS Branch 1 Phones IP Cisco Unified CM Administration Window
Figure 51 Call Routing Class of Control CSS-CSS HQ Phones Analog Cisco Unified CM Administration Window
Figure 52 Call Routing Class of Control CSS-CSS HQ Phones IP Cisco Unified CM Administration Window
Configuring the Cisco Unified CM Media Resources Parameters
Use the Cisco Unified Communications Manager Administration window to configure the media resources parameters. The media resources parameter example configurations are shown in the following sections:
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Media Resources: Annunciator Parameters
To configure the media resources annunciator parameters for the Cisco Unified CM, click Media Resources > Annunciator in the Cisco Unified CM Administration window.
Figure 53 Media Resources Annunciator ANN 2 Cisco Unified CM Administration Window
Media Resources: Conference Bridge Parameters
To configure the media resources conference bridge parameters for the Cisco Unified CM, click Media Resources > Conference Bridge in the Cisco Unified CM Administration window.
Figure 54 Media Resources Conference Bridges Cisco Unified CM Administration Window
Figure 55 Media Resources Conference Bridges CFB Enterprise 1 Branch 1 Cisco Unified CM Administration Window
Figure 56 Media Resources Conference Bridges CFB Enterprise 1 HQ Cisco Unified CM Administration Window
Media Resources: Media Termination Point Parameters
To configure the media resources media termination point parameters for the Cisco Unified CM, click Media Resources > Media Termination Point in the Cisco Unified CM Administration window.
Figure 57 Media Resources Media Termination Point Cisco Unified CM Administration Window
Figure 58 Media Resources Media Termination Point MTP Enterprise 1 Branch 1 Administration Window
Figure 59 Media Resources Media Termination Point MTP Enterprise 1 HQ Cisco Unified CM Administration Window
Media Resources: Music on Hold Server Parameters
To configure the media resources music on hold server parameters for the Cisco Unified CM, click Media Resources > Music On Hold Server in the Cisco Unified CM Administration window.
Figure 60 Media Resources Music on Hold Server MOH Enterprise 1 HQ Cisco Unified CM Administration Window
Media Resources: Transcoder Parameters
To configure the media resources transcoder parameters for the Cisco Unified CM, click Media Resources > Transcoder in the Cisco Unified CM Administration window.
Figure 61 Media Resources Transcoder Cisco Unified CM Administration Window
Figure 62 Media Resources Transcoder XCODE Enterprise 1 Branch 1 Cisco Unified CM Administration Window
Figure 63 Media Resources Transcoder XCODE Enterprise 1 HQ Cisco Unified CM Administration Window
Media Resources: Media Resource Group Parameters
To configure the media resources media resource group parameters for the Cisco Unified CM, click Media Resources > Media Resource Group in the Cisco Unified CM Administration window.
Figure 64 Media Resources-Media Resource Group Cisco Unified CM Administration Window
Figure 65 Media Resources-Media Resource Group Enterprise 1 Branch 1 Cisco Unified CM Administration Window
Figure 66 Media Resources-Media Resource Group Enterprise 1 HQ Cisco Unified CM Administration Window
Media Resources: Media Resource Group List Parameters
To configure the media resources media resource group list parameters for the Cisco Unified CM, click Media Resources > Media Resource Group List in the Cisco Unified CM Administration window.
Figure 67 Media Resources-Media Resource Group List Cisco Unified CM Administration Window
Figure 68 Media Resources-Media Resource Group List Branch 1 HW MRGL Cisco Unified CM Administration Window
Figure 69 Media Resources-Media Resource Group List HQ HW MRGL Cisco Unified CM Administration Window
Configuring the Cisco Unified CM Voice Mail Parameters
Use the Cisco Unified Communications Manager Administration window to configure the voice mail parameters. The voice mail parameter example configurations are shown in the following sections:
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Voice Mail: Cisco Voice Mail Port Parameters
To configure the voice mail Cisco voice mail port parameters for the Cisco Unified CM, click
Voice Mail > Cisco Voice Mail Port in the Cisco Unified CM Administration window.Figure 70 Voice Mail Cisco Voice Mail Port Cisco Unified CM Administration Window
Figure 71 Voice Mail-Voice Mail Port CiscoUM1 VI1 Cisco Unified CM Administration Window
Voice Mail: Message Waiting Parameters
To configure the voice mail message waiting parameters for the Cisco Unified CM, click Voice Mail > Message Waiting in the Cisco Unified CM Administration window.
Figure 72 Voice Mail Message Waiting Cisco Unified CM Administration Window
Figure 73 Voice Mail Message Waiting MWI ON Cisco Unified CM Administration Window
Figure 74 Voice Mail Message Waiting MWI Off Cisco Unified CM Administration Window
Voice Mail: Voice Mail Pilot Parameters
To configure the voice mail voice mail pilot parameters for the Cisco Unified CM, click Voice Mail > Voice Mail Pilot in the Cisco Unified CM Administration window.
Figure 75 Voice Mail-Voice Mail Pilot 1099 Cisco Unified CM Administration Window
Voice Mail: Voice Mail Profile Parameters
To configure the voice mail voice mail profile parameters for the Cisco Unified CM, click Voice Mail > Voice Mail Profile in the Cisco Unified CM Administration window.
Figure 76 Voice Mail-Voice Mail Profile VM Profile Enterprise 1 HQ Cisco Unified CM Administration Window
Configuring the Cisco Unified CM Device Parameters
Use the Cisco Unified Communications Manager Administration window to configure the device parameters. The device parameter example configurations are shown in the following sections:
Device: Gateway Parameters
To configure the device gateway parameters for the Cisco Unified CM, click Device > Gateway in the Cisco Unified CM Administration window.
Figure 77 Device Gateway Cisco Unified CM Administration Window
Figure 78 Device Gateway Enterprise 1 Branch 1 Enterprise 1.com Cisco Unified CM Administration Window
Figure 79 Device Gateway Enterprise 1 Branch 1 Enterprise 1.com pots 1110 Cisco Unified CM Administration Window
Figure 80 Device Gateway Enterprise 1 Branch 1 Enterprise 1.com pots 1110 Line Administration Window
Figure 81 Device Gateway Enterprise 1 HQ VG224 Cisco Unified CM Administration Window
Figure 82 Device Gateway Enterprise 1 HQ VG224 ANA 1050 Cisco Unified CM Administration Window
Figure 83 Device-Gateway Enterprise 1 HQ VG224 ANA 1050 Line Cisco Unified CM Administration Window
Device: Phone Parameters
To configure the device phone parameters for the Cisco Unified CM, click Device > Phone in the Cisco Unified CM Administration window.
Figure 84 Device Phone 4155551000 Cisco Unified CM Administration Window
Figure 85 Device Phone 1000 Cisco Unified CM Administration Window
Figure 86 Device Phone 4155551170 Cisco Unified CM Administration Window
Figure 87 Device Phone 1170 Cisco Unified CM Administration Window
Device: Trunk Parameters
To configure the device trunk parameters for the Cisco Unified CM, click Device > Trunk in the Cisco Unified CM Administration window.
Figure 88 Device Trunk Cisco Unified CM Administration Window
Figure 89 Device Trunk Enterprise 1 HQ CUBE1 Phones Analog Cisco Unified CM Administration Window
Figure 90 Device Trunk Enterprise 1 HQ CUBE1 Phones IP Cisco Unified CM Administration Window
Figure 91 Device Trunk Enterprise 1 Branch 1 CUBE1 Phones Analog Cisco Unified CM Administration Window
Figure 92 Device Trunk Enterprise 1 Branch 1 CUBE1 Phones IP Cisco Unified CM Administration Window
Enterprise 1 HQ Cisco Unity and Cisco Unity Express Example Configuration
To integrate the Cisco Unity version 5.0 with Cisco Unified CM configuration, see Cisco Unified Communications Manager SCCP Integration Guide for Cisco Unity Release 5.0.
Enterprise 1 HQ and Cisco VG224 Analog Phone Gateway Example Configuration
The following example shows a CLI configuration for the enterprise 1 HQ the Cisco VG224 Analog Phone Gateway for the test topology described in Figure 7.
Ent1_HQ_VG224#!stcapp ccm-group 1stcapp!voice service voipfax protocol pass-through g711ulawmodem passthrough nse codec g711ulaw!interface FastEthernet0/0ip address 10.40.97.254 255.255.0.0load-interval 30duplex fullspeed 100!interface FastEthernet0/1no ip addressshutdownduplex autospeed auto!ip forward-protocol ndip route 0.0.0.0 0.0.0.0 FastEthernet0/0!voice-port 2/0timeouts ringing infinitycaller-id enable!voice-port 2/1timeouts ringing infinitycaller-id enable!sccp local FastEthernet0/0sccp ccm 10.40.97.2 identifier 10sccp!sccp ccm group 1associate ccm 10 priority 1!dial-peer voice 1 potsservice stcappport 2/0!dial-peer voice 2 potsservice stcappport 2/1!Ent1_HQ_VG224#Enterprise 1 HQ Cisco ASA Firewall Example Configuration
The following example shows a CLI configuration for the enterprise 1 HQ the Cisco ASA 8.0(4) 5500 Series Adaptive Security Appliances firewall for the test topology described in Figure 7.
Ent1-HQ-ASA#!interface Vlan65nameif insidesecurity-level 100ip address 10.40.99.1 255.255.255.0!interface Vlan70nameif outsidesecurity-level 0ip address 10.40.98.2 255.255.255.0!interface Ethernet0/0description *** To WAN ***switchport access vlan 70!interface Ethernet0/1description *** To LAN ***switchport access vlan 65!ftp mode passiveaccess-list 100 extended permit icmp any anyaccess-list 100 extended permit icmp any any echoaccess-list 100 extended permit icmp any any echo-replyaccess-list 100 extended permit tcp any host 40.40.97.2 eq 2000access-list 100 extended permit udp any host 40.40.97.2 eq sipaccess-list 100 extended permit tcp any host 40.40.97.2 range h323 h323access-list 100 extended permit tcp any host 10.10.11.151 eq 5090access-list 100 extended permit udp any host 10.10.11.151 eq 5090access-list 100 extended permit tcp any host 40.40.97.2 eq 2428access-list 100 extended permit udp any host 40.40.97.2 eq 2427pager lines 24logging enablelogging buffered debugginglogging asdm informationalmtu inside 1500mtu outside 1500icmp unreachable rate-limit 1 burst-size 1asdm image disk0:/asdm-524.binno asdm history enablearp timeout 14400access-group 100 in interface outside!timeout xlate 3:00:00timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolutehttp server enableno snmp-server locationno snmp-server contactsnmp-server enable traps snmp authentication linkup linkdown coldstarttelnet timeout 5ssh timeout 5console timeout 0!class-map sipoutinmatch port udp eq 5090class-map inspection_defaultmatch default-inspection-traffic!policy-map type inspect dns preset_dns_mapparametersmessage-length maximum 512policy-map global_policyclass inspection_defaultinspect dns preset_dns_mapinspect ftpinspect rshinspect rtspinspect esmtpinspect sqlnetinspect skinnyinspect sunrpcinspect xdmcpinspect sipinspect netbiosinspect tftppolicy-map outsideinclass sipoutininspect sipclass inspection_defaultinspect skinny!service-policy global_policy interface insideservice-policy outsidein interface outsideprompt hostname contextCryptochecksum:xxxxxxxxxxxxxxxxxxxxxxxxxxx: endEnt1-HQ-ASA#Branch 1 Cisco UBE, TDM Gateway, and Cisco Unified SRST Example Configuration
The following example shows a CLI configuration for the branch 1 Cisco Unified Border Element, TDM Switching in the Cisco AS5000 Gateway, and Cisco Unified SRST for the test topology described in Figure 7.
Ent1_Br1#!voice-card 4dspfarmdsp services dspfarm!voice service voipaddress-hidingallow-connections sip to sipno supplementary-service sip moved-temporarilyno supplementary-service sip refersupplementary-service media-renegotiatefax protocol pass-through g711ulawmodem passthrough nse codec g711ulawsipmin-se 90header-passing error-passthrumidcall-signaling passthru!voice translation-rule 1rule 1 /^61/ /1/rule 2 /^71/ /1/!voice translation-profile OUTGOING-SIP-TRK-DIGIT-STRIPtranslate called 1!interface Loopback0ip address 10.10.11.154 255.255.255.255!interface GigabitEthernet0/0no ip addressshutduplex autospeed automedia-type rj45!interface GigabitEthernet0/1description *** To Local LAN ***no ip addressip virtual-reassemblyload-interval 30duplex autospeed automedia-type rj45!interface GigabitEthernet0/1.1encapsulation dot1Q 103ip address 10.40.103.1 255.255.255.0ip helper-address 10.40.97.2ip virtual-reassembly!interface Serial4/0:0description *** To WAN ***ip address 10.80.80.82 255.255.255.252ip virtual-reassemblyencapsulation frame-relayload-interval 30cdp enableframe-relay map ip 10.80.80.81 202frame-relay interface-dlci 202no frame-relay inverse-arp NOVELL 202no frame-relay inverse-arp APPLETALK 202no frame-relay inverse-arp DECNET 202frame-relay lmi-type ansiframe-relay local-dlci 202!interface Serial4/0:23no ip addressencapsulation hdlcisdn switch-type primary-net5isdn incoming-voice voiceno cdp enable!call treatment oncall threshold global cpu-avg low 68 high 75call threshold global total-mem low 75 high 85call threshold global total-calls low 1 high 12!!voice-port 2/1/0!voice-port 2/1/1!voice-port 4/0/0!voice-port 4/0/1!voice-port 4/0:23!ccm-manager mgcp!mgcpmgcp call-agent 10.40.97.2 2427 service-type mgcp version 0.1mgcp dtmf-relay voip codec all mode out-of-bandmgcp sdp simplemgcp fax t38 inhibitmgcp bind control source-interface GigabitEthernet0/1.1mgcp bind media source-interface GigabitEthernet0/1.1!mgcp profile default!sccp local GigabitEthernet0/1.1sccp ccm 10.40.97.2 identifier 1 priority 1 version 6.0sccp ip precedence 3sccp!sccp ccm group 1bind interface GigabitEthernet0/1.1associate ccm 1 priority 1associate profile 3 register XCD001AA29DF631associate profile 2 register CON001AA29DF631associate profile 1 register MTP001AA29DF631keepalive retries 1keepalive timeout 10switchover method immediateswitchback method immediate!dspfarm profile 3 transcodedescription transcode bridgecodec g711ulawcodec g729r8maximum sessions 5associate application SCCP!dspfarm profile 2 conferencedescription conference bridgecodec g711ulawcodec g729r8maximum sessions 4associate application SCCP!dspfarm profile 1 mtpcodec g729r8maximum sessions software 5associate application SCCP!!dial-peer voice 2000 voipdescription *** Voice: LAN to WAN - Incoming Dial-Peer ***huntstopcodec g729r8session protocol sipv2incoming called-number 6Tdtmf-relay rtp-nte digit-dropno vad!dial-peer voice 2001 voipdescription *** Voice: LAN to WAN - Outgoing Dial-Peer ***translation-profile outgoing OUTGOING-SIP-TRK-DIGIT-STRIPhuntstopdestination-pattern 6Tcodec g729r8voice-class sip early-offer forcedmax-redirects 5session protocol sipv2session target ipv4:10.3.33.22dtmf-relay rtp-nte digit-dropno vad!dial-peer voice 2100 voipdescription *** Voice: WAN to LAN - Incoming Dial-Peer ***huntstopcodec g729r8session protocol sipv2incoming called-number 415Tdtmf-relay rtp-nte digit-dropno vad!dial-peer voice 2101 voipdescription *** Voice: WAN to LAN - Outgoing Dial-Peer ***huntstopdestination-pattern 415Tcodec g729r8max-redirects 5session protocol sipv2session target ipv4:10.40.97.2dtmf-relay rtp-nte digit-dropno vad!dial-peer voice 3000 voipdescription *** Fax: LAN to WAN - Incoming Dial-Peer ***huntstopsession protocol sipv2incoming called-number 7Tdtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 3001 voipdescription *** Fax: LAN to WAN - Outgoing Dial-Peer ***translation-profile outgoing OUTGOING-SIP-TRK-DIGIT-STRIPhuntstopdestination-pattern 7Tvoice-class sip early-offer forcedmax-redirects 5session protocol sipv2session target ipv4:10.3.33.22dtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 3100 voipdescription *** Fax: WAN to LAN - Incoming Dial-Peer ***huntstopsession protocol sipv2incoming called-number 415555111[0,1]dtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 3101 voipdescription *** Fax: WAN to LAN - Outgoing Dial-Peer ***huntstopdestination-pattern 415555111[0,1]max-redirects 5session protocol sipv2session target ipv4:10.40.97.2dtmf-relay rtp-nte digit-dropcodec g711ulawno vad!dial-peer voice 1 potsservice mgcpappport 4/0/0!dial-peer voice 2 potsservice mgcpappport 4/0/1!dial-peer hunt 3sip-uaauthentication username yyyyy password 7 xxxxxxxxxxno remote-party-idretry invite 2retry response 5retry bye 2retry cancel 2retry register 10retry options 1g729-annexb override!call-manager-fallbackvideomax-conferences 10 gain -6transfer-system full-consultlog table max-size 1000ip source-address 10.40.103.1 port 2000max-ephones 50max-dn 50system message primary Ent1_Br1dialplan-pattern 1 415555.... extension-length 4transfer-pattern .T!Ent1_Br1#Branch 1 Cisco Unity Express 3.2 and Cisco Unified CM Example Configuration
To integrate the Branch 1 Cisco Unity Express with Cisco Unified CM configuration, see
CallManager for Cisco Unity Express Configuration Example.Cisco Unified Border Element Performance Summary
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