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Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x
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Preface
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Introduction
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IP Telephony Deployment Models
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Network Infrastructure
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Gateways
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Cisco Unified CallManager Trunks
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Media Resources
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Music on Hold
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Call Processing
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Call Admission Control
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Dial Plan
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Emergency Services
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Third-Party Voicemail Design
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Cisco Unity
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Cisco Unity Express
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Cisco Unified MeetingPlace Integration
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Cisco Unified MeetingPlace Express
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IP Video Telephony
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LDAP Directory Integration
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IP Telephony Migration Options
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Voice Security
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IP Telephony Endpoints
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Device Mobility
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Recommended Hardware and Software Combinations
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Glossary
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Index
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Table Of Contents
Low-Density Analog Interface Module
High-Density Analog Interface Module
Supported Platforms and Cisco IOS Requirements for Analog Interface Modules
Cisco Communication Media Module (CMM)
WS-X6624-FXS Analog Interface Module
Cisco Unified IP Phone Expansion Module 7914
Maximum IP Communicator Configuration Limits
Maximum Cisco IP SoftPhone Configuration Limits
Codecs Supported by Cisco Unified Video Advantage and Cisco IP Video Phone 7985G
Third-Party SCCP Video Endpoints
Cisco ATA 186 and IP Conference Station
Cisco Unified Wireless IP Phone 7920
Cisco Unified Video Advantage with a Cisco Unified IP Phone
Sony and Tandberg SCCP Endpoints
IP Telephony Endpoints
This chapter summarizes various types of IP Telephony endpoints along with their features and QoS recommendations. The IP Telephony endpoints can be categorized into the following major types:
These sections provide detailed information about each endpoint type. In addition, the section on QoS Recommendations, lists generic QoS configurations, and the Endpoint Features Summary, lists all the endpoint features.
The following list summarizes high-level recommendations for selecting IP Telephony endpoints:
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For low-density analog connections, use the Cisco Analog Telephone Adapter (ATA) or low-density analog interface module.
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Analog Gateways
Analog gateways include router-based analog interface modules, Cisco Communication Media Module (CMM) with 24-FXS port adapter, Catalyst 6500 24-FXS analog interface module, Cisco VG224, Cisco VG248, and Cisco Analog Telephone Adaptor (ATA) 186 and 188. An analog gateway is usually used to connect analog devices, such as fax machines, modems, TDD/TTYs, and analog phones, to the VoIP network so that the analog signal can be packetized and transmitted over the IP network.
Analog Interface Module
Cisco router-based analog interface modules include low-density interface modules (NM-1V, NM-2V, NM-HD-1V, NM-HD-2V, NM-HD-2VE, NM-HDV2, NM-HDV2-1T1/E1, and NM-HDV2-2T1/E1) and high-density interface modules (NM-HDA-4FXS and EVM-HD-8FXS/DID). Cisco analog interface modules connect the PSTN and other legacy telephony equipment, including PBXs, analog telephones, fax machines, and key systems, to Cisco multiservice access routers. Cisco analog interface modules are best suited for connecting low- to high-density analog devices to the IP network with limited call features.
Low-Density Analog Interface Module
The low-density analog interface modules include the NM-1V, NM-2V, NM-HD-1V, NM-HD-2V, NM-HD-2VE, NM-HDV2, NM-HDV2-1T1/E1, and NM-HDV2-2T1/E1. The NM-1V and NM-2V contain one or two interface cards (VIC). The interface cards include: two-port FXS VIC (VIC-2FXS); two-port FXO VIC (VIC-2FXO, VIC-2FXO-M1/M2/M3, and VIC-2FXO-EU); two-port Direct Inward Dial VIC (VIC-2DID); two-port E&M VIC (VIC-2E/M); two-port Centralized Automated Message Accounting VIC (VIC-2CAMA); and two-port BRI VIC (VIC-2BRI-S/T-TE and VIC-2BRI-NT/TE). The NM-1V and NM-2V can serve up to two and four FXS connections, respectively.
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The NM-HD-1V and NM-HD-2V contain one and two VICs, respectively. The NM-HD-2VE contains two VICs or two voice/WAN interface cards (VWIC), or a combination of one VIC and one VWIC. The NM-HD-1V, NM-HD-2V, and NM-HD-2VE can serve up to 4, 8, and 8 FXS or FXO connections, respectively. The NM-HDV2, NM-HDV2-1T1/E1, and NM-HDV2-2T1/E1 can be fitted with either digital T1/E1 or analog/BRI voice interface cards, with up to 4 FXS or FXO connections. The difference among these three interface modules is that the NM-HDV2-1T1/E1 has one built-in T1/E1 port while the NM-HDV2-2T1/E1 has two built-in T1/E1 ports.
The voice interface cards include: 2-port and 4-port FXS VICs (VIC2-2FXS and VIC-4FXS/DID); 2-port and 4-port FXO VICs (VIC2-2FXO and VIC2-4FXO); 2-port Direct Inward Dial VIC (VIC-2DID); 2-port E&M VIC (VIC2-2E/M); and 2-port BRI VIC (VIC2-2BRI-NT/TE). The voice/WAN interface cards include: 1-port and 2-port RJ-48 multiflex trunk (MFT) T1/E1 VWICs for both voice and WAN connections (VWIC-1MFT-T1, VWIC-2MFT-T1, VWIC-2MFT-T1-DI, VWIC-1MFT-E1, VWIC-2MFT-E1, VWIC-2MFT-E1-DI, VWIC-1MFT-G703, VWIC-2MFT-G703, VWIC2-1MFT-T1/E1, VWIC2-2MFT-T1/E1, VWIC2-1MFT-G703, and VWIC2-2MFT-G703). The G.703 interface cards are primarily for data connectivity but can in some cases be configured to support voice applications.
High-Density Analog Interface Module
The high-density analog interface module includes the NM-HDA-4FXS and EVM-HD-8FXS/DID. The NM-HDA-4FXS has four on-board FXS ports and room for two expansion modules from the following options:
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The NM-HDA-4FXS provides up to 12 analog ports (4 FXS and 8 FXO) with four built-in FXS ports and two EM-HDA-4FXO or EM2-HDA-4FXO extension modules, or 16 analog ports (12 FXS and 4 FXO) with four built-in FXS ports and one EM-HDA-8FXS, and one EM-HDA-4FXO or EM2-HDA-4FXO extension module. A configuration using two 8-port FXS extension modules is not supported. The NM-HDA also has a connector for a daughter module (DSP-HDA-16) that provides additional DSP resources to serve an additional 8 high-complexity calls or 16 medium-complexity calls.
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The EVM-HD-8FXS/DID provides eight individual ports on the baseboard module and can be configured for FXS or DID signaling. In addition, the EVM-HD-8FXS/DID has room for two expansion modules from the following options:
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These extension modules can be used in any combination and provide for configurations of up to 24 FXS ports per EVM-HD-8FXS/DID.
Supported Platforms and Cisco IOS Requirements for Analog Interface Modules
The supported platforms for Cisco analog interface modules are the Cisco 2600, 2800, 3600, 3700, and 3800 Series. Table 21-1 lists the maximum number of interface modules supported per platform, and Table 21-2 lists the minimum Cisco IOS software version required.
Cisco Communication Media Module (CMM)
The Cisco CMM is a line card that provides high-density analog, T1, and E1 gateway connections for Catalyst 6000 and Cisco 7600 Series switches. The Cisco CMM can serve up to 72 FXS connections. The CMM operates as either an MGCP or H.323 gateway, and it provides Survivable Remote Site Telephony (SRST) service for up to 480 IP phones.
Cisco CMM can contain the following interface port adapters: 24-port FXS analog port adapter (WS-SVC-CMM-24FXS), 6-port T1 interface port adapter (WS-SVC-CMM-6T1), 6-port E1 interface port adapter (WS-SVC-CMM-6E1), and conference/transcoding port adapter (WS-SVC-CMM-ACT). Table 21-3 lists the minimum software requirements for the compatible port adapters.
WS-X6624-FXS Analog Interface Module
The Cisco WS-X6624-FXS analog interface module is an MGCP-based device for connecting high-density analog devices to the IP telephony network, and it provides 24 analog ports.
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Cisco VG224 Gateway
The Cisco VG224 analog gateway is a Cisco IOS high-density 24-port gateway for connecting analog devices to the IP Telephony network. In Cisco IOS Release 12.4(2)T and later, the Cisco VG224 can act as a Skinny Client Control Protocol (SCCP), Media Gateway Control Protocol (MGCP), or H.323 endpoint with Cisco Unified CallManager and re-home to a Survivable Remote Site Telephone (SRST) router in failover scenarios. The Cisco VG224 supports Cisco Unified CallManager Release 3.1 and later. The Cisco VG224 also supports modem pass-through, modem relay, fax pass-through, and fax relay.
Cisco VG248 Gateway
The Cisco VG248 is a high-density, 48 port, Skinny Client Control Protocol (SCCP) gateway for connecting analog devices such as analog phones, fax machines, modems and speakerphones to an enterprise Cisco Unified CallManager (Release 3.1 and later) and voice network. The Cisco VG248 also supports Cisco Unified CallManager integration with legacy voicemail systems and PBXs compatible with Simplified Message Desk Interface (SMDI), NEC Message Center Interface (MCI), or Ericsson voicemail protocols. The Cisco VG248 supports failover to Survivable Remote Site Telephone (SRST).
Cisco ATA 186 and 188
The Cisco Analog Telephone Adaptor (ATA) 186 or 188 can connect two analog devices to the IP telephony network, and it is the best suited for low-density analog devices connecting to the IP network.
The difference between the Cisco ATA 186 and 188 is that the former has only one 10 Base-T Ethernet connection while the later has an integrated Ethernet switch providing two 10/100 Base-T Ethernet connections for itself and a co-located PC or other Ethernet-based device. Cisco Unified CallManager 4.2 has native SCCP support for the Cisco ATA 186 or 188.
The Cisco ATA 186 and 188 can be configured in any of the following ways:
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Cisco Unified IP Phones
The Cisco IP phone portfolio includes basic IP phones, business IP phones, manager IP Phones, and executive IP phones. For a complete list of supported features of Cisco basic IP phone models, see the Endpoint Features Summary.
Cisco Basic IP Phones
The Cisco basic IP phone is best suited for low-traffic users with limited call features and budget requirements. The Cisco basic IP phones include Cisco Unified IP Phone 7902G, 7905G, 7906G, 7911G, and 7912G.
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Cisco Business IP Phones
The Cisco business IP phone is best suited for the transaction-type worker with medium telephony traffic use and extensive call features, such as speakers, headset, and so forth. The business IP phones include Cisco Unified IP Phone 7940G, 7941G, and 7941G-GE.
Cisco Manager IP Phones
The Cisco manager IP phone is best suited for managers and administrative assistants with medium to heavy telephony traffic use and extensive call features such as speakers, headset, and so forth. The business IP phones include Cisco Unified IP Phone 7960G, 7961G, and 7961G-GE.
Cisco Executive IP Phones
The Cisco executive IP phone is best suited for the executive high-traffic user with extensive call features. The executive IP phones include the Cisco Unified IP Phone 7970G and 7971G-GE.
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Cisco Unified IP Phone Expansion Module 7914
The Cisco Unified IP Phone Expansion Module 7914 is for administrative assistants and others who need to determine the status of a number of lines beyond the current line capability of the phone.
The Cisco Unified IP Phone Expansion Module 7914 extends the capability of the Cisco Unified IP Phone 7960G, 7961G, 7961G-GE, 7970G, or 7971G-GE with additional buttons and an LCD. The Cisco Unified IP Phone Expansion Module 7914 provides 14 buttons per module, and the Cisco Unified IP Phones 796xG and 797xG can support up to two Cisco Unified IP Phone Expansion Modules. If the IP phone uses Cisco inline power or IEEE802.3af PoE, then the Cisco Unified IP Phone Expansion Module 7914 requires the use of an external power adaptor (CP-PWR-CUBE-3).
Software-Based Endpoints
Software-based endpoints include Cisco IP Communicator and Cisco IP SoftPhone. A software-based endpoint is an application installed on a client PC, and it registers with (and is controlled by) Cisco Unified CallManager.
Cisco IP Communicator
Cisco IP Communicator is a Microsoft Windows-based application that endows computers with the functionality of IP phones. This application enables high-quality voice calls on the road, in the office, or from wherever users can access the corporate network. It is an ideal solution for remote users and telecommuters. Cisco IP Communicator is easy to deploy and features some of the latest technology and advancements available with IP communications today. This section summarizes the following design considerations that apply when using Cisco IP Communicator with Cisco Unified CallManager:
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Maximum IP Communicator Configuration Limits
Because Cisco IP Communicator is an SCCP standalone device, the design guidelines for IP phones in the various IP Telephony deployment models still hold true for the Cisco IP Communicator. Refer to the chapter on IP Telephony Deployment Models, page 2-1, for details.
Codec Selection
The Cisco IP Communicator supports G.711 and G.729a codecs. The codec selection can be done by configuring the region in which Cisco IP Communicator is located. Cisco recommends G.729a low-bandwidth codec configurations in deployments with telecommuters connecting their Cisco IP Communicator across the WAN. The Cisco IP Communicator also has a low-bandwidth codec overriding capability in a G.711 region, which can be enabled by checking the Optimize for Low Bandwidth option in the Audio setting window (see Figure 21-1). In this case, the Cisco IP Communicator will use a G.729 codec to set up a call with another phone in the same region. Cisco IP Communicator can make video calls when it is used together with Cisco Unified Video Advantage. For more information on the video codec support of Cisco Unified Video Advantage, see the section on Video Endpoints.
Figure 21-1 Cisco IP Communicator Audio Setting
Call Admission Control
Call admission control ensures that there is enough bandwidth available to process IP phone calls over the network. While there are several mechanisms for implementing call admission control, the Cisco IP Communicator uses the locations mechanism configured in Cisco Unified CallManager for centralized call processing deployments. Refer to the chapter on Call Admission Control, page 9-1, for more information on call admission control with Cisco Unified CallManager locations.
With Cisco Unified CallManager 4.2, the Cisco IP Communicator can automatically update its location setting as it moves from one location to another. Cisco Unified CallManager performs the call admission control based on the device's current location setting and subtracts a certain amount of bandwidth for the call from the current location's bandwidth pool. For more information on device mobility, refer to the chapter on Device Mobility, page 22-1.
Cisco IP SoftPhone
This section summarizes the following design considerations that apply when using Cisco IP SoftPhone with Cisco Unified CallManager:
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The information in this section applies explicitly to Cisco IP SoftPhone Release 1.3. For specific details about Cisco IP SoftPhone configuration and features, refer to the Cisco IP Softphone Administrator Guide (1.3), available online at
http://www.cisco.com/en/US/products/sw/voicesw/ps1860/tsd_products_support_eol_series_home.html
Figure 21-2 illustrates that the Cisco IP SoftPhone application can either monitor or control the hardware IP phone associated with it. In the case of a third-party controlled phone, the Cisco IP SoftPhone acts as a virtual extension of the desktop IP phone. The Cisco IP SoftPhone application is able to see and handle incoming and outgoing calls for the hardware phone. From the perspective of provisioning devices and CTI resources, each user with this configuration would be provisioned as a third-party controlled IP phone. As a CTI port, the Cisco IP SoftPhone is a dedicated line to process calls directly to the client machine without the additional control and monitoring of a desktop phone.
Figure 21-2 Cisco IP SoftPhone Device Association Options
The Cisco IP SoftPhone is not able to run a CTI port and a third-party controlled phone with the same directory number (DN) at the same time. As shown in Figure 21-2, the user is able to have extension 8110 as either a CTI port or a control for the desktop phone.
For more information on device and resource provisioning, refer to the chapter on Call Processing, page 8-1.
Maximum Cisco IP SoftPhone Configuration Limits
Regardless of the device limits allowed per server, there are maximum limits on the number CTI devices you can configure in Cisco Unified CallManager. The CTI device limits as they apply to Cisco IP SoftPhone are as follows:
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The following assumptions apply to the preceding maximum Cisco IP SoftPhone limits:
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Codec Selection
Cisco IP SoftPhone supports G.711, G.723, and G.729a codecs. Cisco recommends G.729a low-bandwidth codec configurations in deployments with telecommuters connecting their Cisco IP SoftPhones across the WAN.
Because Cisco Unified CallManager does not support the G.723 codec, Cisco IP Softphone has two bandwidth codec settings to use: G.711 is the default setting, with a user-configurable option to select the low-bandwidth codec setting of G.729 on the TAPI Service Provider (TSP) client (see Figure 21-3). Refer to the chapter on Network Infrastructure, page 3-1, for details on provisioning network bandwidth.
Figure 21-3 Cisco IP Softphone Audio Setting
Cisco IP SoftPhone users with low-bandwidth connections across the WAN should consider selecting this low-bandwidth G.729 codec setting.
Call Admission Control
Call admission control ensures that there is enough bandwidth available to process IP phone calls over the network. While there are several mechanisms for implementing call admission control, the Cisco IP SoftPhone uses the locations mechanism configured in Cisco Unified CallManager for centralized call processing deployments. Refer to the chapter on Call Admission Control, page 9-1, for more information on call admission control with Cisco Unified CallManager locations.
With Cisco Unified CallManager 4.2, the Cisco IP SoftPhone can automatically update its location setting as it moves from one location to another. Cisco Unified CallManager performs the call admission control based on the device's current location setting and subtracts a certain amount of bandwidth for the call from the current location's bandwidth pool. For more information on device mobility, refer to the chapter on Device Mobility, page 22-1.
Wireless Endpoints
Cisco wireless endpoints use a wireless LAN (WLAN) infrastructure via wireless access points (APs) to provide telephony functionality and features. This type of endpoint is ideal for environments with the need for mobile users within a area where traditional wired phones are undesirable or problematic. (Refer to Wireless LAN Infrastructure, page 3-58, for more information about wireless network design.)
The Cisco Unified Wireless IP Phone 7920 is a hardware-based phone with a built-in radio antenna that enables 802.11b wireless LAN connectivity to the network. These phones register with Cisco Unified CallManager using Skinny Client Control Protocol (SCCP), just like the hardware-based phones and Cisco IP Communicator. For more information, refer to the Cisco Unified Wireless IP Phone 7920 Design and Deployment Guide, available at
Site Survey
Before deploying the Cisco Unified Wireless IP Phone 7920, you must perform a complete site survey to determine the appropriate number and location of APs required to provide radio frequency (RF) coverage. Your site survey should take into consideration which types of antennas will provide the best coverage, as well as where sources of RF interference might exist. A site survey requires the use of the Site Survey tool on the Cisco Unified Wireless IP Phone 7920 (accessed via Menu > Network Config > Site Survey) and the Aironet Client Utility Site Survey Tool used with a Cisco Aironet NIC card on a laptop or PC. Additional third-party tools can also be used for site surveys; however, Cisco highly recommends that you conduct a final site survey using the Cisco Unified Wireless IP Phone 7920 because each endpoint or client radio can behave differently depending on antenna sensitivity and survey application limitations.
Authentication
To connect to the wireless network, the Cisco Unified Wireless IP Phone 7920 must first use one of the following authentication methods to associate and communicate with the AP:
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This method allows the Cisco Unified Wireless IP Phone 7920 to be authenticated to the AP via 802.1X with a user name and password once a secure authenticated tunnel is established between the client and an EAP-compliant Remote Authentication, Authorization, and Accounting server via Protected Access Credential (PAC). Upon authentication, traffic to and from the wireless device is encrypted using TKIP or WEP. Using the 802.1X authentication method and the PAC authenticated tunnel exchange requires an EAP-compliant Remote Authentication Dial-In User Service (RADIUS) authentication server such as the Cisco Secure Access Control Server (ACS), which provides access to a user database.
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This method allows the Cisco Unified Wireless IP Phone 7920 to be authenticated to the AP via 802.1X with a user name and password. Upon authentication, traffic to and from the wireless device is encrypted using Temporal Key Integrity Protocol (TKIP). Using the 802.1X authentication method requires an EAP-compliant Remote Authentication Dial-In User Service (RADIUS) authentication server such as the Cisco Secure Access Control Server (ACS), which provides access to a user database.
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This method allows the Cisco Unified Wireless IP Phone 7920 to be authenticated to the AP via the configuration of a shared key on the Cisco Unified Wireless IP Phone 7920 and the AP. Upon authentication, traffic to and from the wireless device is encrypted using TKIP. This method of authentication is not recommended for enterprise deployments.
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This method allows the Cisco Unified Wireless IP Phone 7920 to be authenticated to the AP via 802.1x with a user name and password. Upon authentication, traffic to and from the wireless device is encrypted using either WEP 128 or TKIP. The 802.1X authentication method requires an EAP-compliant RADIUS authentication server such as the Cisco ACS, which provides access to a user database for the initial authentication request. Subsequent authentication requests are validated via the wireless domain service (WDS) at the AP, which shortens re-authentication times and ensures fast, secure roaming.
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This method allows the Cisco Unified Wireless IP Phone 7920 and AP to be authenticated mutually based on a user name and password. Upon authentication, the dynamic key is generated and used for encrypting traffic between the Cisco Unified Wireless IP Phone 7920 and the AP. A LEAP-compliant Radius authentication server, such as the Cisco Secure Access Control Server (ACS), is required to provide access to the user database.
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This method involves the configuration of static 10 (40-bit) or 26 (128-bit) character keys on the Cisco Unified Wireless IP Phone 7920 and the AP. This method is AP-based authentication in which access to the network is gained if the device has a matching key.
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This method requires no exchange of identifying information between the Cisco Wireless IP Phone 7920 and the AP. Cisco does not recommend this method because it provides no secure exchange of voice or signaling, and it allows any rouge device to associate to the AP.
Capacity
Each AP can support a maximum of seven active G.711 voice streams or eight G.729 streams. If these numbers are exceeded, poor quality can result due to dropped or delayed voice packets or dropped calls. AP rates set lower than 11 Mbps will result in lower call capacity per AP.
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Based on these active call capacity limits, and using Erlang ratios, you can calculate the number of Cisco Unified Wireless IP Phone 7920s that each AP can support. For example, given a typical user-to-call capacity ratio of 3:1, a single AP can support 21 to 24 Cisco Unified Wireless IP Phone 7920s, depending on whether the codec used is G.711 or G.729. However, this number does not take into consideration the possibility that other Cisco Unified Wireless IP Phone 7920s could roam to this AP, so a lower number of phones per AP is more realistic.
The number of APs per VLAN or Layer 2 subnet should also be considered. To optimize memory and performance on the APs, Cisco recommends deploying no more than 30 APs on a single VLAN or subnet. This recommendation, taken with typical user-to-call capacity ratios, limits the number of Cisco Unified Wireless IP Phone 7920s per Layer 2 subnet to approximately 500 (or about 15 to 17 Cisco Unified Wireless IP Phone 7920s per AP).
These capacities were calculated with voice activity detection (VAD) disabled and a packetization sample size of 20 milliseconds (ms). VAD is a mechanism for conserving bandwidth by not sending RTP packets while no speech is occurring during the call. However, enabling or disabling VAD is a global cluster-wide configuration parameter on Cisco Unified CallManager. (It is referred to as Silence Suppression in Cisco Unified CallManager.) Thus, if VAD is enabled for the Cisco Unified Wireless IP Phone 7920, then it will be enabled for all devices in the Cisco Unified CallManager cluster. Cisco recommends leaving VAD (Silence Suppression) disabled to provide better overall voice quality.
At a sampling rate of 20 ms, a voice call will generate 50 packets per second (pps) in either direction. Cisco recommends setting the sample rate to 20 ms for almost all cases. By using a larger sample size (for example, 30 or 40 ms), you can increase the number of simultaneous calls per AP, but a larger end-to-end delay will result. In addition, the percentage of acceptable voice packet loss within a wireless environment decreases dramatically with a larger sample size because more of the conversation is missing when a packet is lost. For more information about voice sampling size, see Bandwidth Provisioning, page 3-44.
Phone Configuration
You can configure the Cisco Unified Wireless IP Phone 7920 either through the phone's keypad or with the 7920 Configuration Utility running on a PC that is attached to the phone via a USB cable. In either case, you must configure the following parameters:
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Configure either the DHCP server address or static settings such as IP address, subnet mask, default gateway, TFTP server, and DNS server, as appropriate for the network. These settings can be found on the Cisco Unified Wireless IP Phone 7920 under Menu > Profiles > Network Profile.
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Configure the service set identifier (SSID) for the voice VLAN and the authentication type, including the WEP key and/or LEAP user name and password when appropriate. These settings can be found on the Cisco Unified Wireless IP Phone 7920 under Menu > Profiles > Network Profile.
Roaming
Currently the Cisco Unified Wireless IP Phone 7920 is able to roam at Layer 2 (within the same VLAN or subnet) and still maintain an active call.
Layer 2 roaming occurs in the following situations:
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In trying to determine eligible AP roam targets for Layer 2 roaming, the wireless IP phone uses the following variables to determine the best AP to associate with:
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Used by the wireless IP phone to determine the signal strength and quality of available APs within an RF coverage area. The phone will attempt to associate with the AP that has the highest RSSI value and matching authentication/encryption type.
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Enables the AP to communicate channel utilization information to the wireless phone. The phone will use the QBSS value to determine if it should attempt to roam to another AP, because APs with high channel utilization might not be able to handle VoIP traffic effectively.
Layer 2 roaming times for the wireless IP phone depend on the type of authentication used. If static WEP keys are used for authentication between the phone and the AP, Layer 2 roaming occurs in less than 100 ms. If LEAP (with local Cisco Secure ACS authentication) is used, Layer 2 roaming occurs in 200 to 400 ms. Using Cisco Centralized Key Management (Cisco CKM) can reduce roaming time to less than 100 ms.
When devices roam at Layer 3, they move from one AP to another AP across native VLAN boundaries. The Cisco Catalyst 6500 Series Wireless LAN Services Module (WLSM) allows the Cisco Unified Wireless IP Phone 7920 to roam at Layer 3 while still maintaining an active call. The Cisco Wireless IP Phone 7920 can roam at Layer 3 by using Static WEP or Cisco CKM protocols. Cisco CKM enables the phone to achieve full Layer 3 mobility while using either WEP 128 or TKIP encryption. Seamless Layer 3 roaming occurs only when the client is roaming within the same mobility group. For details about the Cisco WLSM and Layer 3 roaming, refer to the product documentation available at
If you are using 802.1x authentication in the wireless LAN, Cisco CKM is recommended to minimize roaming downtime. Whether the roaming is at Layer 2 or Layer 3, device downtime decreases from 300-400 ms to 100 ms or less. Cisco CKM also takes some of the load off the ACS server by reducing the number of authentication requests that must be sent to the ACS.
AP Call Admission Control
Call admission control mechanisms in Cisco Unified CallManager or in a gatekeeper can control WAN bandwidth utilization and provide QoS for existing calls, but both mechanisms are applied only at the beginning of a call. For calls between static devices, this type of call admission control is sufficient. However, for a call between two mobile wireless devices such the Cisco Unified Wireless IP Phone 7920, there must also be a call admission control mechanism at the AP level because the wireless devices may roam from one AP to another.
The AP mechanism for call admission control is QBSS, which is the beacon information element that enables the AP to communicate channel utilization information to the wireless IP phone. As previously mentioned, this QBSS value helps the phone determine whether it should roam to another AP. A lower QBSS value indicates that the AP is a good candidate to roam to, while a higher QBSS value indicates that the phone should not roam to this AP.
While this QBSS information is useful, it does not guarantee that calls will retain proper QoS during roaming. When a Cisco Unified Wireless IP Phone 7920 is associated to an AP with a high QBSS, the AP will prevent a call from being initiated or received by rejecting the call setup and sending a Network Busy message to the initiating phone. However, once a call is set up between a wireless IP phone and another endpoint, the phone may roam and associate with an AP with a high QBSS, thus resulting in an oversubscription of the available bandwidth on that AP.
Cisco IP Conference Station
The Cisco IP Conference Station combines conference room speaker-phone technology with Cisco IP Communications technology. The Cisco IP Conference Station is best suited for use in conferencing environments providing 360-degree room coverage.
The Cisco Unified IP Conference Station 7936 has an external speaker and three built-in microphones. The Cisco Unified IP Conference Station 7936 requires Cisco CallManager Release 3.3(3) SR3 or later. The Cisco Unified IP Conference Station 7936 also features a pixel-based LCD display with backlighting, and optional extension microphones can be connected to it for extended microphone coverage in larger rooms.
Video Endpoints
Cisco Unified CallManager Release 4.2 supports the following types of video-enabled endpoints:
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SCCP Video Endpoints
SCCP video endpoints register directly with Cisco Unified CallManager and download their configurations via Trivial File Transfer Protocol (TFTP). They support many features and supplementary services, including hold, transfer, conference, park, pickup and group pickup, music on hold, shared line appearances, mappable softkeys, call forwarding (busy, no answer, and unconditional), and much more.
Cisco Unified Video Advantage
Cisco Unified Video Advantage is a Windows-based application and USB camera that you can install on a Windows 2000 or Windows XP personal computer. When the PC is physically connected to the PC port on a Cisco Unified IP Phone 7940, 7941, 7960, 7961, 7970, or 7971 running the Skinny Client Control Protocol, the Cisco Unified Video Advantage application "associates" with the phone, thus enabling users to operate their phones as they always have but now with the added benefit of video. In Cisco Unified Video Advantage Release 2.0, this association can also be to Cisco IP Communicator running on the same PC.
The system administrator can control which IP Phones allow this association to take place by toggling the Video Capabilities: Enabled/Disabled setting on the IP Phone configuration page in Cisco Unified CallManager Administration. When this feature is enabled, an icon representing a camera appears in the bottom right-hand corner of the IP Phone display. By default, Cisco Unified Video Advantage is disabled. You can also use the Bulk Administration Tool to modify this setting on many phones at once. Note that the PC Port: Enabled/Disabled setting must also be enabled for Cisco Unified Video Advantage to work with a hardware IP Phone; however, the PC Access to Voice VLAN setting does not have to be enabled.
To achieve the association with a hardware IP Phone, Cisco Unified Video Advantage installs a Cisco Discovery Protocol (CDP) driver onto the Ethernet interface of the PC. CDP enables the PC and the hardware IP Phone to discover each other automatically, which means that the user does not have to configure anything on the PC or the hardware IP Phone in order for Cisco Unified Video Advantage to work. The user can, therefore, plug the PC into any hardware IP Phone that is video-enabled and automatically associates with it. (See Figure 21-4.)
Cisco Unified Video Advantage 2.0 does not rely on CDP to discover the presence of Cisco IP Communicator running on the same PC. Instead, it listens for a private Windows message sent from the IP Communicator process. Once IP Communicator is discovered, the association process works exactly like it does for a hardware IP Phone. (See Figure 21-5.)
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Figure 21-4 Cisco Unified Video Advantage Operational Overview
Figure 21-4 illustrates the following events:
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Figure 21-5 Cisco IP Communicator Associating with Cisco Unified Video Advantage
Figure 21-5 illustrates the following events:
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When a call is made using Cisco Unified Video Advantage, the audio is handled by the IP Phone while the video is handled by the PC. There is no synchronization mechanism between the two devices, so QoS is essential to minimize jitter, latency, fragmented packets, and out-of-order packets.
If a hardware IP Phone is used, the IP Phone resides in the voice VLAN, while the PC resides in the data VLAN, which means that there must be a Layer-3 routing path between the voice and data VLANs in order for the association to occur. If there are access control lists (ACLs) or firewalls between these VLANs, they must be configured to permit the association protocol (which uses TCP port 4224 in both directions) to pass. If Cisco IP Communicator is used, this communication happens internal to the PC, and there are no Layer-3 boundaries to cross.
Cisco Unified Video Advantage supports Differentiated Services Code Point (DSCP) traffic classifications. Cisco Unified CallManager specifies the DSCP value in the SCCP messages it sends to the phone. When the IP Phone makes an audio-only call, it marks its SCCP control traffic as DSCP CS3 and its audio RTP media traffic as DSCP EF. However, when the IP Phone makes a video call, it marks its SCCP control traffic as DSCP CS3 and its audio RTP media traffic as DSCP AF41, and the Cisco Unified Video Advantage application marks its video RTP media traffic as DSCP AF41 as well. Both the IP Phone and the Cisco Unified Video Advantage application mark their "association" protocol messages as DSCP CS3 because it is considered to be signaling traffic and is grouped with all other signaling traffic such as SCCP.
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Cisco Unified Video Advantage, like any other application that runs on a PC, does have an impact on system performance, which you should take into consideration. Cisco Unified Video Advantage 1.0 supports two types of video codecs: H.263 and the Cisco VT Camera Wideband Video Codec. Cisco Unified Video Advantage 2.0 also supports two types of codecs: H.263 and H.264. The Cisco VT Camera Wideband Video Codec places the least demand on the PC but the most demand on the network. H.263 places a lesser demand on the network but a higher demand on the PC. Finally, H.264 places the least demand on the network but the highest demand on the PC. Therefore, if your network has plenty of available bandwidth, you can use the Cisco VT Camera Wideband Video Codec and save on PC CPU and memory resources.
The H.263 and H.264 codecs support a range of speeds up to 1.5 Mbps. In summary, customers must balance PC performance with network utilization when deploying Cisco Unified Video Advantage.
System Requirements
For detailed PC requirements, refer to the Cisco Unified Video Advantage Data Sheet, available at
http://www.cisco.com/warp/public/cc/pd/nemnsw/callmn/prodlit/vtadv_ds.htm
Cisco IP Video Phone 7985G
The Cisco IP Video Phone 7985G is a personal desktop video phone. Unlike Cisco Unified Video Advantage, which is an application that runs on a PC, the Cisco IP Video Phone 7985G is a standalone phone with integrated video features. The phone has an 8.4 inch color LCD screen and an embedded video camera for making video calls. The phone supports up to eight line appearances, has two 10/100 BaseT Ethernet connections, and has buttons for Directories, Messages, Settings, and Services. Like other Cisco Unified IP Phones, the Cisco IP Video Phone 7985G uses CDP to learn VLAN and CoS information from the attached switch to use in 802.1p/q markings.
Codecs Supported by Cisco Unified Video Advantage and Cisco IP Video Phone 7985G
Table 21-4 lists the codecs supported by Cisco Unified Video Advantage and Cisco IP Video Phone 7985G
Third-Party SCCP Video Endpoints
Two manufacturers of video endpoints, Sony and Tandberg, currently have the following products that support the Cisco Skinny Client Control Protocol (SCCP):
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SCCP on both the Sony and Tandberg endpoints is modeled after SCCP on the Cisco Unified IP Phone 7940. Most features found on the Cisco Unified IP Phone 7940 user interface are also supported on the Sony endpoints as well as the Tandberg endpoints, including multiple line appearances, softkeys, and buttons for Directories, Messages, Settings, Services, and so forth. The Sony and Tandberg endpoints also support the Option 150 field in DHCP to discover the IP address of the TFTP server, and they download their configurations from the TFTP server. However, software upgrades of the Sony and Tandberg endpoints are not done via TFTP. Instead, the customer must manually upgrade each endpoint using tools provided by the vendor. (Tandberg uses an FTP method, while Sony uses FTP or a physical memory stick.) The Sony and Tandberg endpoints register with up to three Cisco Unified CallManager servers and will fail-over to its secondary or tertiary servers if its primary server becomes unreachable.
While the Sony and Tandberg endpoints support softkey functionality similar to that of the Cisco Unified IP Phone 7940 and 7960, the exact feature support differs between vendor and model. Check the manufacturers' documentation for supported features. Features that are currently known to be missing on some platforms include:
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Because the Sony and Tandberg endpoints use SCCP, dialing a video call from an endpoint is similar to dialing an audio call from a Cisco Unified IP Phone. If users are familiar with Cisco Unified IP Phones, they should also find the Sony and Tandberg endpoints very intuitive to use. The main difference in the user interface is that the Sony and Tandberg endpoints do not have a button keypad or a handset like those on a phone. Instead, the remote control is used to access features and to dial numbers on the Sony and Tandberg endpoints.
Note
Codecs Supported by Sony and Tandberg SCCP Endpoints
Codec support for the third-party SCCP endpoints varies by vendor, model, and software version. Check the vendors' product documentation for the supported codecs.
QoS Recommendations
This sections provides the basic QoS guidelines and configurations for the Cisco Catalyst switches most commonly deployed with IP Telephony endpoints. For more details, refer to the Quality of Service design guide at
Cisco VG224 and VG248
Analog gateways are trusted endpoints. For Cisco VG224 and VG248 gateways, configure the switch to trust the DSCP value of the VG248 packets. The following sections list the commands to configure the most common Cisco Catalyst switches for the Cisco VG224 and VG248 analog gateways.
Note
Cisco 2950
CAT2950(config)#interface interface-idCAT2950(config-if)#mls qos trust dscpCAT2950(config-if)#switchport mode accessCAT2950(config-if)#switchport access vlan vvlan_id
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Cisco 2970 or 3750
CAT2970(config)#mls qosCAT2970(config)interface interface-idCAT2970(config-if)#mls qos trust dscpCAT2970(config-if)#switchport mode accessCAT2970(config-if)#switchport access vlan vvlan_idCisco 3550
CAT3550(config)#mls qosCAT3550(config)interface interface-idCAT3550(config-if)#mls qos trust dscpCAT3550(config-if)#switchport mode accessCat3550(config-if)#switchport access vlan vvlan_idCisco 4500 with SUPIII, IV, or V
CAT4500(config)#qosCAT4500(config)interface interface-idCAT4500(config-if)#qos trust dscpCAT4500(config-if)#switchport mode accessCAT4500(config-if)#switchport access vlan vvlan_idCisco 6500
CAT6500>(enable)set qos enableCAT6500>(enable)set port qos 2/1 vlan-basedCAT6500>(enable)set vlan vvlan_id mod/portCAT6500>(enable)set port qos mod/port trust trust-dscpCisco ATA 186 and IP Conference Station
Because the Cisco Analog Telephone Adaptor (ATA) 186 and IP Conference Station are trusted endpoints, their QoS configurations are identical to those described in the section on Cisco VG224 and VG248.
Cisco ATA 188 and IP Phones
For the Cisco Analog Telephone Adaptor (ATA) 188 and IP Phones, Cisco recommends segregating the voice VLAN from the data VLAN. For the Cisco ATA 186, 7902, 7905, 7906, 7910, and IP Conference Station, Cisco still recommends configuring voice and data VLAN segregation and an auxiliary voice VLAN. In this way, the same access-layer configurations can be used with different IP phone models and ATAs, and end-users can plug their IP phones or ATAs into different access ports on the switch and get the same treatment. For the Cisco ATA 186, 7902, 7905, 7906, 7910, and IP Conference Stations, the command to override the CoS value of the frames from the attached PC has no effects because these devices do not have a PC connected to them.
The following sections list the configuration commands for IP phones on the most commonly deployed Cisco Catalyst switches.
Cisco 2950
CAT2950(config)#CAT2950(config)#class-map VVLANCAT2950(config-cmap)# match access-group name VVLANCAT2950(config-cmap)#class-map VLANCAT2950(config-cmap)# match access-group name DVLANCAT2950(config-cmap)#exitCAT2950(config)#CAT2950(config)#policy-map IPPHONE-PCCAT2950(config-pmap)# class VVLANCAT2950(config-pmap-c0# set ip dscp 46CAT2950(config-pmap-c)# police 1000000 8192 exceed-action-dropCAT2950(config-pmap)# class DVLANCAT2950(config-pmap-c0# set ip dscp 0CAT2950(config-pmap-c)# police 5000000 8192 exceed-action-dropCAT2950(config-pmap-c)#exitCAT2950(config-pmap)#exitCAT2950(config)#CAT2950(config)#interface interface-idCAT2950(config-if)#mls qos trust device cisco-phoneCAT2950(config-if)#mls qos trust cosCAT2950(config-if)#switchport mode accessCAT2950(config-if)#switchport voice vlan vvlan_idCAT2950(config-if)#switchport access vlan dvlan_idCAT2950(config-if)#service-policy input IPPHONE-PCCAT2950(config-if)#exitCAT2950(config)#CAT2950(config)#ip access-list standard VVLANCAT2950(config-std-nacl)# permit voice_IP_subnet wild_card_maskCAT2950(config-std-nacl)#exitCAT2950(config)#ip access-list standard DVLANCAT2950(config-std-nacl)# permit data_IP_subnet wild_card_maskCAT2950(config-std-nacl)#end
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Cisco 2970 or 3750
CAT2970(config)# mls qos map cos-dscp 0 8 16 24 34 46 48 56CAT2970(config)# mls qos map policed-dscp 0 24 to 8CAT2970(config)#CAT2970(config)#class-map match-all VVLAN-VOICECAT2970(config-cmap)# match access-group name VVLAN-VOICECAT2970(config-cmap)#CAT2970(config-cmap)#class-map match-all VVLAN-CALL-SIGNALINGCAT2970(config-cmap)# match access-group name VVLAN-CALL-SIGNALINGCAT2970(config-cmap)#CAT2970(config-cmap)#class-map match-all VVLAN-ANYCAT2970(config-cmap)# match access-group name VVLAN-ANYCAT2970(config-cmap)#CAT2970(config-cmap)# policy-map IPPHONE-PCCAT2970(config-pmap)#class VVLAN-VOICECAT2970(config-pmap-c)# set ip dscp 46CAT2970(config-pmap-c)# police 128000 8000 exceed-action dropCAT2970(config-pmap-c)# class VVLAN-CALL-SIGNALINGCAT2970(config-pmap-c)# set ip dscp 24CAT2970(config-pmap-c)# police 32000 8000 exceed-action policed-dscp-transmitCAT2970(config-pmap-c)# class VVLAN-ANYCAT2970(config-pmap-c)# set ip dscp 0CAT2970(config-pmap-c)# police 32000 8000 exceed-action policed-dscp-transmitCAT2970(config-pmap-c)# class class-defaultCAT2970(config-pmap-c)# set ip dscp 0CAT2970(config-pmap-c)# police 5000000 8000 exceed-action policed-dscp-transmitCAT2970(config-pmap-c)# exitCAT2970(config-pmap)# exitCAT2970(config)#CAT2970(config)#CAT2970(config)#interface interface-idCAT2970(config-if)# switchport voice vlan vvlan_idCAT2970(config-if)# switchport access vlan dvlan_idCAT2970(config-if)# mls qos trust device cisco-phoneCAT2970(config-if)# service-policy input IPPHONE-PCCAT2970(config-if)# exitCAT2970(config)#CAT2970(confiig)#ip access list extended VVLAN-VOICECAT2970(config-ext-nacl)# permit udp Voice_IP_Subnet Subnet_Mask any range 16384 32767 dscp efCAT2970(config-ext-nacl)# exitCAT2970(config)#ip access list extended VVLAN-CALL-SIGNALINGCAT2970(config-ext-nacl)# permit tcp Voice_IP_Subnet Subnet_Mask any range 2000 2002 dscp cs3CAT2970(config-ext-nacl)# permit tcp Voice_IP_Subnet Subnet_Mask any range 2000 2002 dscp Af31CAT2970(config-ext-nacl)# exitCAT2970(config)#ip access list extended VVLAN-ANYCAT2970(config-ext-nacl)# permit ip Voice_IP_Subnet Subnet_Mask anyCAT2970(config-ext-nacl)# endCAT2970#Cisco 3550
CAT3550(config)# mls qos map cos-dscp 0 8 16 24 34 46 48 56CAT3550(config)# mls qos map policed-dscp 0 24 26 46 to 8CAT3550(config)#class-map match-all VOICECAT3550(config-cmap)# match ip dscp 46CAT3550(config-cmap)#class-map match-any CALL SIGNALINGCAT3550(config-cmap)# match ip dscp 26CAT3550(config-cmap)# match ip dscp 24CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all VVLAN-VOICECAT3550(config-cmap)# match vlan vvlan_idCAT3550(config-cmap)# match class-map VOICECAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all VVLAN-CALL-SIGNALINGCAT3550(config-cmap)# match vlan vvlan_idCAT3550(config-cmap)# match class-map CALL SIGNALINGCAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all ANYCAT3550(config-cmap)# match access-group name ACL_NameCAT3550(config-cmap)#CAT3550(config-cmap)# class-map match-all VVLAN-ANYCAT3550(config-cmap)# match vlan vvlan_idCAT3550(config-cmap)# match class-map ANYCAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all DVLAN-ANYCAT3550(config-cmap)# match vlan dvlan_idCAT3550(config-cmap)# match class-map ANYCAT3550(config-cmap)#CAT3550(config-cmap)#policy-map IPPHONE-PCCAT3550(config-pmap)# class VVLAN-VOICECAT3550(config-pmap-c)# set ip dscp 46CAT3550(config-pmap-c)# police 128000 8000 exceed-action dropCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class VVLAN-CALL-SIGNALINGCAT3550(config-pmap-c)# set ip dscp 24CAT3550(config-pmap-c)# police 32000 8000 exceed-action policed-dscp-transmitCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class VVLAN-ANYCAT3550(config-pmap-c)# set ip dscp 0CAT3550(config-pmap-c)# police 32000 8000 exceed-action policed-dscp-transmitCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class DVLAN-VOICECAT3550(config-pmap-c)# set ip dscp 0CAT3550(config-pmap-c)# police 5000000 8000 exceed-action policed-dscp-transmitCAT3550(config-pmap-c)#exitCAT3550(config-pmap)#exitCAT3550(config)#interface interface-idCAT3550(config-if)# switchport voice vlan vvlan_idCAT3550(config-if)# switchport access vlan dvlan_idCAT3550(config-if)# mls qos trust device cisco-phoneCAT3550(config-if)# service-policy input IPPHONE-PCCAT3550(config-if)# exitCAT3550(config)#CAT3550(confiig)#ip access list standard ACL_ANYCAT3550(config-std-nacl)# permit anyCAT3550(config-std-nacl)# endCAT3550#Cisco 4500 with SUPIII, IV, or V
CAT4500(config)# qos map cos 5 to dscp 46CAT4500(config)# qos map cos 0 24 26 46 to dscp 8CAT4500(config)#CAT4500(config)#class-map match-all VVLAN-VOICECAT4500(config-cmap)# match access-group name VVLAN-VOICECAT4500(config-cmap)#CAT4500(config-cmap)#class-map match-all VVLAN-CALL-SIGNALINGCAT4500(config-cmap)# match access-group name VVLAN-CALL-SIGNALINGCAT4500(config-cmap)#CAT4500(config-cmap)#class-map match-all VVLAN-ANYCAT4500(config-cmap)# match access-group name VVLAN-ANYCAT4500(config-cmap)#CAT4500(config-cmap)# policy-map IPPHONE-PCCAT4500(config-pmap)#class VVLAN-VOICECAT4500(config-pmap-c)# set ip dscp 46CAT4500(config-pmap-c)# police 128 kps 8000 byte exceed-action dropCAT4500(config-pmap-c)# class VVLAN-CALL-SIGNALINGCAT4500(config-pmap-c)# set ip dscp 24CAT4500(config-pmap-c)# police 32 kps 8000 byte exceed-action policed-dscp-transmitCAT4500(config-pmap-c)# class VVLAN-ANYCAT4500(config-pmap-c)# set ip dscp 0CAT4500(config-pmap-c)# police 32 kps 8000 byte exceed-action policed-dscp-transmitCAT4500(config-pmap-c)# class class-defaultCAT4500(config-pmap-c)# set ip dscp 0CAT4500(config-pmap-c)# police 5 mpbs 8000 byte exceed-action policed-dscp-transmitCAT4500(config-pmap-c)# exitCAT4500(config-pmap)# exitCAT4500(config)#CAT4500(config)#CAT4500(config)#interface interface-idCAT4500(config-if)# switchport voice vlan vvlan_idCAT4500(config-if)# switchport access vlan dvlan_idCAT4500(config-if)# qos trust device cisco-phoneCAT4500(config-if)# service-policy input IPPHONE-PCCAT4500(config-if)# exitCAT4500(config)#CAT4500(confiig)#ip access list extended VVLAN-VOICECAT4500(config-ext-nacl)# permit udp Voice_IP_Subnet Subnet_Mask any range 16384 32767 dscp efCAT4500(config-ext-nacl)# exitCAT4500(config)#ip access list extended VVLAN-CALL-SIGNALINGCAT4500(config-ext-nacl)# permit tcp Voice_IP_Subnet Subnet_Mask any range 2000 2002 dscp cs3CAT4500(config-ext-nacl)# permit tcp Voice_IP_Subnet Subnet_Mask any range 2000 2002 dscp Af31CAT4500(config-ext-nacl)# exitCAT4500(config)#ip access list extended VVLAN-ANYCAT4500(config-ext-nacl)# permit ip Voice_IP_Subnet Subnet_Mask anyCAT4500(config-ext-nacl)# endCAT4500#Cisco 6500
CAT6500> (enable) set qos cos-dscp-map 0 8 16 24 32 46 48 56CAT6500> (enable) set qos policed-dscp-map 0, 24, 26, 46:8CAT6500> (enable)CAT6500> (enable) set qos policer aggregate VVLAN-VOICE rate 128 burst 8000 dropCAT6500> (enable) set qos policer aggregate VVLAN-CALL-SIGNALING rate 32 burst 8000 policed-dscpCAT6500> (enable) set qos policer aggregate VVLAN-ANY rate 5000 burst 8000 policed-dscpCAT6500> (enable) set qos policer aggregate PC-DATA rate 5000 burst 8000 policed-dscpCAT6500> (enable)CAT6500> (enable) set qos acl ip IPPHONE-PC dscp 46 aggregate VVLAN-VOICE udp Voice_IP_Subnet Subnet_Mask any range 16384 32767CAT6500> (enable) set qos acl ip IPPHONE-PC dscp 24 aggregate VVLAN-CALL-SIGNALING tcp Voice_IP_Subnet Subnet_Mask any range 2000 2002CAT6500> (enable) set qos acl ip IPPHONE-PC dscp 0 aggregate VVLAN-ANY Voice_IP_Subnet Subnet_Mask anyCAT6500> (enable) set qos acl ip IPPHONE-PC dscp 0 aggregate PC-DATA anyCAT6500> (enable) commit qos acl IPPHONE-PCCAT6500> (enable) set vlan vvlan_id mod/portCAT6500> (enable) set port qos mod/port trust-device ciscoipphoneCAT6500> (enable) set qos acl map IPPHONE-PC mod/portCAT6500> (enable)Software-Based Endpoints
Even though Cisco IP Communicator and IP SoftPhone mark their signaling packets and media packets respectively, Cisco recommends that you re-mark the DSCP value of the packets from the PC that is running Cisco IP Communicator or IP SoftPhone because the PC is not a trusted device in the network. Instead of using the full range of User Datagram Protocol (UDP) ports (16384 to 32767) for the media packets, Cisco IP Communicator and IP SoftPhone can be configured to use a specific UDP port.
Cisco Unified Video Advantage 2.0 brings video functionality to Cisco IP Communicator, therefore Cisco IP Communicator is able to make video calls. Cisco recommends that you configure an access control list (ACL) to match the video packets from Cisco Unified Video Advantage on UDP port 5445, and re-mark the traffic with DSCP value AF41. Unlike Cisco Unified IP Phones, Cisco IP Communicator does not automatically mark its voice packets with DSCP value AF41 when a video call is made; instead, it marks its voice packets with DSCP value EF. Hence, the voice stream of a video call will be marked with DSCP value EF while the video stream of the call will be marked with DSCP value AF41.
For Cisco IP Communicator, you can specify the UDP port by using one of the following options:
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If you use both options to set the UDP port and port range, settings from the second option will override the first option.
For Cisco IP SoftPhone, you can specify the UDP port and port range under Network Audio Settings > Audio Output Port.
The following sections list the QoS configuration commands for Cisco IP Communicator and IP SoftPhone on the most commonly deployed Cisco Catalyst switches.
Cisco 2950 with Enhanced Image
The Cisco Catalyst 2950 Series switches are not recommended for software-based endpoint QoS implementations due to two limitations:
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Cisco 2970 or 3750
CAT2970 (config) #mls qosCAT2970 (config) #mls qos map policed-dscp 0 24 26 46 to 8CAT2970 (config) #CAT2970 (config) #class-map match-all COMMUNICATOR-VOICECAT2970 (config-cmap) # match access-group name COMMUNICATOR-VOICECAT2970 (config-cmap) #class-map match-all COMMUNICATOR-VIDEOCAT2970 (config-cmap) # match access-group name COMMUNICATOR-VIDEOCAT2970 (config-cmap) #class-map match-all COMMUNICATOR-SIGNALINGCAT2970 (config-cmap) # match access-group name COMMUNICATOR-SIGNALINGCAT2970 (config-cmap) #exitCAT2970 (config) #CAT2970 (config) #policy-map COMMUNICATOR-PCCAT2970 (config-pmap-c) #class COMMUNICATOR-VOICECAT2970 (config-pmap-c) # set ip dscp 46CAT2970 (config-pmap-c) # police 128000 8000 exceed-action dropCAT2970 (config-pmap-c) #class COMMUNICATOR-VIDEOCAT2970 (config-pmap-c) # set ip dscp 34CAT2970 (config-pmap-c) # police 50000000 8000 exceed-action policed-dscp-transmitCAT2970 (config-pmap-c) #class COMMUNICATOR-SIGNALINGCAT2970 (config-pmap-c) # set ip dscp 24CAT2970 (config-pmap-c) # police 32000 8000 exceed-action policed-dscp-transmitCAT2970 (config-pmap-c) #class class-defaultCAT2970 (config-pmap-c) # set ip dscp 0CAT2970 (config-pmap-c) # police 5000000 8000 exceed-action policed-dscp transmitCAT2970 (config-pmap-c) # exitCAT2970 (config-pmap) #exitCAT2970 (config) #CAT2970 (config) #interface FastEthernet interface-idCAT2970 (config-if) # switchport access vlan vvlan_idCAT2970 (config-if) # switchport mode accessCAT2970 (config-if) # service-policy input COMMUNICATOR-PCCAT2970 (config-if) # exitCAT2970 (config) #ip access list extended COMMUNICATOR-VOICECAT2970 (config-ext-nacl) # permit udp host PC_IP_address eq fixed_port_number anyCAT2970 (config-ext-nacl) # exitCAT2970 (config) #ip access list extended COMMUNICATOR-VIDEOCAT2970 (config-ext-nacl) # permit udp host PC_IP_address any eq 5445CAT2970 (config-ext-nacl) # exitCAT2970(config)#ip access-list extended COMMUNICATOR-SIGNALINGCAT2970(config-ext-nacl)# permit tcp host PC_IP_address host CallManager_IP_address eq 2748 or 2000CAT2970(config-ext-nacl)# exitCisco 3550
CAT3550 (config) #mls qosCAT3550 (config) #mls qos map cos-dscp 0 8 16 24 34 46 48 56CAT3550 (config) #mls qos map policed-dscp 0 24 26 34 46 to 8CAT3550 (config) #CAT3550 (config) #class-map match-all COMMUNICATOR-VOICECAT3550 (config-cmap) # match access-group name COMMUNICATOR-VOICECAT3550 (config-cmap) #class-map match-all COMMUNICATOR-VIDEOCAT3550 (config-cmap) # match access-group name COMMUNICATOR-VIDEOCAT3550 (config-cmap) #class-map match-all COMMUNICATOR-SIGNALINGCAT3550 (config-cmap) # match access-group name COMMUNICATOR-SIGNALINGCAT3550 (config-cmap) #exitCAT3550 (config) #CAT3550 (config) #policy-map COMMUNICATOR -PCCAT3550 (config-pmap-c) #class COMMUNICATOR -VOICECAT3550 (config-pmap-c) # set ip dscp 46CAT3550 (config-pmap-c) # police 128000 8000 exceed-action dropCAT3550 (config-pmap-c) #class COMMUNICATOR-VIDEOCAT3550 (config-pmap-c) # set ip dscp 34CAT3550 (config-pmap-c) # police 50000000 8000 exceed-action policed-dscp-transmitCAT3550 (config-pmap-c) #class COMMUNICATOR-SIGNALINGCAT3550 (config-pmap-c) # set ip dscp 24CAT3550 (config-pmap-c) # police 32000 8000 exceed-action policed-dscp-transmitCAT3550 (config-pmap-c) #class class-defaultCAT3550 (config-pmap-c) # set ip dscp 0CAT3550 (config-pmap-c) # police 5000000 8000 exceed-action policed-dscp transmitCAT3550 (config-pmap-c) # exitCAT3550 (config-pmap) #exitCAT3550 (config) #CAT3550 (config) #interface FastEthernet interface-idCAT3550 (config-if) # switchport access vlan vvlan_idCAT3550 (config-if) # switchport mode accessCAT3550 (config-if) # service-policy input COMMUNICATOR-PCCAT3550 (config-if) # exitCAT3550 (config) #ip access list extended COMMUNICATOR-VOICECAT3550 (config-ext-nacl) # permit udp host PC_IP_address eq fixed_port_number anyCAT3550 (config-ext-nacl) # exitCAT2970 (config) #ip access list extended COMMUNICATOR-VIDEOCAT3550 (config-ext-nacl) # permit udp host PC_IP_address any eq 5445CAT3550 (config-ext-nacl) # exitCAT3550(config)#ip access-list extended COMMUNICATOR-SIGNALINGCAT3550(config-ext-nacl)# permit tcp host PC_IP_address host CallManager_IP_address eq 2748 or 2000CAT3550(config-ext-nacl)# exitCisco 4500 with SUPIII, IV, or V
CAT4500(config) #qosCAT4500(config) #qos map policed-dscp 0 24 26 34 46 to 8CAT4500(config) #CAT4500(config) #class-map match-all COMMUNICATOR-VOICECAT4500(config-cmap) # match access-group name COMMUNICATOR-VOICECAT4500(config-cmap) #class-map match-all COMMUNICATOR-VIDEOCAT4500(config-cmap) # match access-group name COMMUNICATOR-VIDEOCAT4500(config-cmap) #class-map match-all COMMUNICATOR-SIGNALINGCAT4500(config-cmap) # match access-group name COMMUNICATOR-SIGNALINGCAT4500(config-cmap) #exitCAT4500(config) #CAT4500(config) #policy-map COMMUNICATOR-PCCAT4500(config-pmap-c) #class COMMUNICATOR-VOICECAT4500(config-pmap-c) # set ip dscp EFCAT4500(config-pmap-c) # police 128 kps 8000 byte exceed-action dropCAT4500(config-pmap-c) #class COMMUNICATOR-VIDEOCAT4500(config-pmap-c) # set ip dscp AF41CAT4500(config-pmap-c) # police 5 mbps 8000 byte exceed-action dropCAT4500(config-pmap-c) #class COMMUNICATOR-SIGNALINGCAT4500(config-pmap-c) # set ip dscp CS3CAT4500(config-pmap-c) # police 32000 kps 8000 byte exceed-action policed-dscp-transmitCAT4500(config-pmap-c) #class class-defaultCAT4500(config-pmap-c) # set ip dscp defaultCAT4500(config-pmap-c) # police 5 mbps 8000 byte exceed-action policed-dscp transmitCAT4500(config-pmap-c) # exitCAT4500(config-pmap) #exitCAT4500(config) #CAT4500(config) #interface FastEthernet interface-idCAT4500(config-if) # switchport access vlan vvlan_idCAT4500(config-if) # switchport mode accessCAT4500(config-if) # service-policy input COMMUNICATOR-PCCAT4500(config-if) # exitCAT4500(config) #ip access list extended COMMUNICATOR-VOICECAT4500(config-ext-nacl) # permit udp host PC_IP_address eq fixed_port_number anyCAT4500(config-ext-nacl) # exitCAT4500(config) #ip access list extended COMMUNICATOR-VIDEOCAT4500(config-ext-nacl) # permit udp host PC_IP_address any eq 5445CAT4500(config-ext-nacl) # exitCAT4500(config)#ip access-list extended COMMUNICATOR-SIGNALINGCAT4500(config-ext-nacl)# permit tcp host PC_IP_address host CallManager_IP_address eq 2748 or 2000CAT4500(config-ext-nacl)# exitCisco 6500
CAT6500> (enable) set qos enableCAT6500> (enable) set qos policed-dscp-map 0, 24, 26, 34, 46:8CAT6500> (enable)CAT6500> (enable) set qos policer aggregate COMMUNICATOR-VOICE rate 128 burst 8000 dropCAT6500> (enable) set qos policer aggregate COMMUNICATOR-VIDEO rate 5000 burst 8000 policed-dscpCAT6500> (enable) set qos policer aggregate COMMUNICATOR-SIGNALING rate 32 burst 8000 policed-dscpCAT6500> (enable) set qos policer aggregate PC-DATA rate 5000 burst 8000 policed-dscpCAT6500> (enable)CAT6500> (enable) set qos acl ip COMMUNICATOR-PC dscp 46 aggregate COMMUNICATOR-VOICE udp host PC_IP_address eq fixed_port_number anyCAT6500> (enable) set qos acl ip COMMUNICATOR-PC dscp 34 aggregate COMMUNICATOR-VIDEO udp host PC_IP_address any eq 5445CAT6500> (enable) set qos acl ip COMMUNICATOR-PC dscp 24 aggregate COMMUNICATOR-SIGNALING tcp host PC_IP_address host CallManager_IP_address eq 2748 or 2000CAT6500> (enable) set qos acl ip COMMUNICATOR-PC dscp 0 aggregate PC-DATA anyCAT6500> (enable) commit qos acl COMMUNICATOR-PCCAT6500> (enable) set vlan vvlan_id mod/portCAT6500> (enable) set port qos mod/port trust untrustedCAT6500> (enable) set qos acl map COMMUNICATOR-PC mod/portCAT6500> (enable)
Note
Cisco Unified Wireless IP Phone 7920
By default, the Cisco Unified Wireless IP Phone 7920 marks its SCCP signaling messages using a Per-Hop Behavior (PHB) value of CS3 or a Differentiated Services Code Point (DSCP) value of 24 (this corresponds to a ToS value of 0x60), and it marks RTP voice packets using a PHB value of EF or a DSCP value of 46 (ToS of 0xB8). With proper queueing on the AP and configuration on the upstream first-hop switch to trust the AP's port, the wireless IP phone traffic will receive the same treatment as wired IP phone traffic. This practice allows the QoS settings to be consistent from LAN to WLAN environments.
In addition, the Cisco Unified Wireless IP Phone 7920 will automatically announce its presence to the AP using the Cisco Discovery Protocol (CDP). The CDP packets are sent from the wireless IP phone to the AP, and they identify the phone so that the AP can place all traffic to the phone in the high-priority queue.
While Ethernet switch ports can typically transmit and receive at 100 Mbps, 802.11b APs have a lower throughput rate that allows for a maximum data rate of 11 Mbps. Furthermore, wireless LANs are a shared medium and, due to contention for this medium, the actual throughput is substantially lower. This throughput mismatch means that, with a burst of traffic, the AP will drop packets, thus adding excessive processor burden to the unit and affecting performance.
By taking advantage of the policing and rate limiting capabilities of the Cisco Catalyst 3550 and 6500 Series switches, you can eliminate the need for the AP to drop excessive packets by configuring the upstream switch port to rate-limit or police traffic going to the AP. The switch port configurations in the following sections rate-limit the port(s) to a practical throughput of 7 Mbps for 802.11b and guarantee 1 Mbps for high-priority voice and control traffic. Furthermore, as indicated in the configuration examples, packets coming from the AP should be trusted and, based on the VLAN tag of each packet, the DSCP marking should either be maintained or marked down. Thus, packets sourced from the Cisco Unified Wireless IP Phone 7920 on the voice VLAN should maintain the appropriate DSCP marking, and packets source from data devices on the data VLAN should be remarked to a DSCP value of 0.
Cisco 3550
CAT3550(config)#mls qosCAT3550(config)#mls qos map cos-dscp 0 8 16 24 32 46 48 56CAT3550(config)#mls qos map policed-dscp 24 46 to 8CAT3550(config)#mls qos aggregate-policer AGG-POL-1M-VOICE-OUT 1000000 8000 exceed-action policed-dscp-transmitCAT3550(config)#mls qos aggregate-policer AGG-POL-6M-DEFAULT-OUT 6000000 8000 exceed-action dropCAT3550(config)#CAT3550(config)#class-map match-all EGRESS-DSCP-0CAT3550(config-cmap)#match ip dscp 0CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all EGRESS-DSCP-8CAT3550(config-cmap)#match ip dscp 8CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all EGRESS-DSCP-16CAT3550(config-cmap)#match ip dscp 16CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all EGRESS-DSCP-32CAT3550(config-cmap)#match ip dscp 32CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all EGRESS-DSCP-48CAT3550(config-cmap)#match ip dscp 48CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all EGRESS-DSCP-56CAT3550(config-cmap)#match ip dscp 56CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all VOICE-SIGNALINGCAT3550(config-cmap)#match ip dscp 24CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all VOICECAT3550(config-cmap)#match ip dscp 46CAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all INGRESS-DATACAT3550(config-cmap)#match anyCAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all INGRESS-VVLAN-VOICECAT3550(config-cmap)#match vlan vvlan-idCAT3550(config-cmap)#match class-map VOICECAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all INGRESS-VVLAN-VOICE-SIGNALINGCAT3550(config-cmap)#match vlan vvlan-idCAT3550(config-cmap)#match class-map VOICE-SIGNALINGCAT3550(config-cmap)#CAT3550(config-cmap)#class-map match-all INGRESS-DVLANCAT3550(config-cmap)#match vlan dvlan-idCAT3550(config-cmap)#match class-map INGRESS-DATACAT3550(config-cmap)#CAT3550(config-cmap)#policy-map EGRESS-RATE-LIMITERCAT3550(config-pmap)#class EGRESS-DSCP-0CAT3550(config-pmap-c)#police aggregate AGG-POL-6M-DEFAULT-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class EGRESS-DSCP-8CAT3550(config-pmap-c)#police aggregate AGG-POL-6M-DEFAULT-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class EGRESS-DSCP-16CAT3550(config-pmap-c)#police aggregate AGG-POL-6M-DEFAULT-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class EGRESS-DSCP-32CAT3550(config-pmap-c)#police aggregate AGG-POL-6M-DEFAULT-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class EGRESS-DSCP-48CAT3550(config-pmap-c)#police aggregate AGG-POL-6M-DEFAULT-OUTCAT3550(config-pmap-c)#class EGRESS-DSCP-56CAT3550(config-pmap-c)#police aggregate AGG-POL-6M-DEFAULT-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class EGRESS-VOICECAT3550(config-pmap-c)#police aggregate AGG-POL-1M-VOICE-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class EGRESS-VOICE-SIGNALINGCAT3550(config-pmap-c)#police aggregate AGG-POL-1M-VOICE-OUTCAT3550(config-pmap-c)#CAT3550(config-pmap-c)#policy-map INGRESS-QOSCAT3550(config-pmap)#class INGRESS-VVLAN-VOICECAT3550(config-pmap-c)#set ip dscp 46CAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class INGRESS-VVLAN-CALL-SIGNALINGCAT3550(config-pmap-c)#set ip dscp 24CAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class INGRESS-DVLANCAT3550(config-pmap-c)#set ip dscp 0CAT3550(config-pmap-c)#CAT3550(config-pmap-c)#class class-defaultCAT3550(config-pmap-c)#set ip dscp 0CAT3550(config-pmap-c)#CAT3550(config-pmap-c)#interface interface idCAT3550(config-if)#description 11Mb towards Wireless Access PointCAT3550(config-if)#switchport access dvlan-idCAT3550(config-if)#switchport voice vvlan-idCAT3550(config-if)#mls qos trust dscpCAT3550(config-if)#service-policy output EGRESS-RATE-LIMITERCAT3550(config-if)#service-policy input INGRESS-QOSCisco 6500
CAT6500> (enable) set qos enableCAT6500> (enable) set qos cos-dscp-map 0 8 16 24 32 46 48 56CAT6500> (enable) set qos policed-dscp-map 24,46:8CAT6500> (enable)CAT6500> (enable) set qos policer microflow VOICE-OUT rate 1000 burst 32 policed-dscpCAT6500> (enable) set qos policer microflow DATA-OUT rate 6000 burst 32 dropCAT6500> (enable)CAT6500> (enable) set qos acl ip AP-VOICE-EGRESS dscp 24 microflow VOICE-OUT ip any any dscp-field 24CAT6500> (enable) set qos acl ip AP-VOICE-EGRESS dscp 46 microflow VOICE-OUT ip any any dscp-field 46CAT6500> (enable) set qos acl ip AP-DATA-EGRESS dscp 0 microflow DATA-OUT ip any anyCAT6500> (enable)CAT6500> (enable) set qos acl ip AP-VOICE-INGRESS trust-dscp ip any anyCAT6500> (enable) set qos acl ip AP-DATA-INGRESS dscp 0 ip any anyCAT6500> (enable)CAT6500> (enable) set qos acl map AP-VOICE-EGRESS vvlan-id outputCAT6500> (enable) set qos acl map AP-DATA-EGRESS dvlan-id outputCAT6500> (enable) set qos acl map AP-VOICE-INGRESS vvlan-id inputCAT6500> (enable) set qos acl map AP-DATA-INGRESS dvlan-id inputCAT6500> (enable)CAT6500> (enable) set port qos mod/port vlan-basedCAT6500> (enable)CAT6500> (enable) set port qos mod/port trust trust-dscpCAT6500> (enable)Video Telephony Endpoints
This section discusses how the following types of endpoint devices classify traffic:
•
•
Cisco Unified Video Advantage with a Cisco Unified IP Phone
The Cisco Unified Video Advantage application residing on the user's PC supports the classification of video packets using DSCP and, therefore, only at Layer 3. The current best practices for Cisco Unified Communications design recommend that the upstream Ethernet switch to which the phone is attached should be configured to trust the 802.1p CoS from the phone. Because the PC packets are unlikely to have an 802.1Q tag, they are unable to support 802.1p CoS bits. This lack of 802.1p support from the PC leaves the following possible options for providing QoS for Cisco Unified Video Advantage:
Option 1
If your current QoS model extends trust to the IP Phone, then the voice and signaling packets will be correctly marked as they ingress the network. With an additional ACL on the port to match UDP port 5445, the video media channel will also be classified to PHB AF41. Without this ACL, the video media would be classified Best Effort and would incur poor image quality and lip-sync issues. The same ACL could also be used to match the CAST connection between the Cisco Unified Video Advantage PC and the IP Phone, which uses TCP port 4224 (classifying it as CS3), although the benefit of doing so is minimal. The signaling packets from the PC, which is on the data VLAN, are returned over the same high-speed port onto the voice VLAN, therefore they are highly unlikely to encounter any congestion.
Option 2
The Enterprise QoS Solution Reference Network Design Guide, Version 3.1 (available at http://www.cisco.com/go/srnd) presents another method. This alternative method recommends changing the port to trust the DSCP of incoming traffic instead of trusting CoS, and then running the incoming packets through a series of Per-Port/Per-VLAN Access Control Lists that match packets based on their TCP/UDP ports (along with other criteria) and police them to appropriate levels. For instance, Cisco Unified Video Advantage will mark its video packets with DSCP AF41, with the switch port set to trust DSCP. The packet will run through an ACL that matches it based on the fact that it is using UDP port 5445, is marked with DSCP AF41, and is coming in on the data VLAN. This ACL will then be used in a class map or policy map to trust the DSCP and police the traffic to N kbps (where N is the amount of video bandwidth you want to allow per port). Similar ACLs and policers will be present for the voice and signaling packets from the IP Phone in the voice VLAN.
Note
Cisco IP Video Phone 7985G
Like many other Cisco Unified IP Phones, the Cisco IP Video Phone 7985G supports 802.1p/Q tagging for traffic originating from the phone and, because the Cisco IP Video Phone 7985G has a second Ethernet interface for PC access, traffic originating from attached devices as well. The current best practices for Cisco Unified Communications design recommend that the upstream Ethernet switch to which the phone is attached should be configured to trust the 802.1p CoS from the phone. Cisco recommends that trust not be extended to the PC port of the phone and, if the switch supports it, that you configure policers to limit the maximum amount of voice, video, and signaling traffic.
Sony and Tandberg SCCP Endpoints
Sony and Tandberg SCCP endpoints correctly mark their media and signaling packets at Layer 3 using DSCP. They do not, however, support 802.1Q and are therefore unable to classify using 802.1p CoS. If you use the UDP and TCP port-matching option, you would be able to classify the SCCP signaling correctly as CS3 and the video media as AF41; however, you would be unable to tell when a UDP port is being used in a voice-only call and should therefore be classified as EF. In such a case, the call admission control mechanisms would not be able to account for the bandwidth correctly. To avoid this situation, there is only one viable option for how to classify and trust traffic from a Sony or Tandberg endpoint:
Option 1
Trust DSCP on the port used by the Sony or Tandberg endpoint. If the switch allows it, configure policers to limit the maximum amount of EF, AF41, and CS3 traffic that can be received on that port. Any other device plugged into that port should not necessarily be trusted, even if its packets are classified using DSCP. This option may be acceptable if the Sony or Tandberg system is a permanent installation in an office or small conference room.
Because the Sony or Tandberg device does not support CDP, the VLAN placement of this endpoint requires manual modification if the requirement is to place it in the voice VLAN. The advantage of placing the endpoint directly in the voice VLAN is that it can be treated like any other IP Telephony endpoint in the system. The disadvantage is that the port might pose a security risk because it provides direct access to the voice VLAN. Alternatively, you can leave the Sony or Tandberg endpoint in the data VLAN, but you will have to provision access between the data and voice VLANs to permit SCCP signaling to Cisco Unified CallManager and to allow the UDP media streams to pass between the data and voice VLANs during voice or video calls.
H.323 Video Endpoints
This type of endpoint is potentially the most challenging from a QoS perspective due to the wide range of H.323 video endpoints, the variation in implementations, and the feature sets. There are two main QoS options for these endpoints; the first relies on the H.323 video endpoint to correctly mark all the traffic, and the second relies on detailed knowledge of the TCP and UDP ports used.
Option 1
If the endpoint correctly marks the media and signaling traffic (signaling should include H.225, H.245, and RAS), you could trust the classifications. Because it is unlikely that the endpoint supports 802.1Q (and therefore 802.1p CoS), you will probably have to use IP Precedence or DSCP in this case. The choice of classification type depends on the specific vendor, model, and software version.
Note
Option 2
Using a combination of either source, destination, or both TCP and UDP port numbers (possibly including IP addresses as well), you could define an ACL that matches and classifies the traffic correctly. In addition, Cisco recommends that you also apply policers to limit the amount of each class of traffic that is admitted to the network. This option has the same potential as Option 1 for classifying voice-only calls incorrectly.
Endpoint Features Summary
The following tables summarize the features supported by the various endpoint devices discussed in this chapter:
•
•
•
•
•
Table 21-5 Cisco Analog Gateway Features
cmm-24fxs
fxsEthernet Connection
N
N
N
Y1
Y2
Y3
Maximum number of Analog Ports
244
72
24
24
48
2
Caller ID
Y
N
N
Y
Y
Y
Call Waiting
N
N
N
N
Y
Y
Caller ID on Call Waiting
N
N
N
N
Y
Y
Call Hold
N
N
N
Y5
N
Y
Call Transfer
N
N
N
Y5
Y
Y
Call Forward
N
N
N
N
Y6
Y
Auto-Answer
N
N
N
N
N
N
Ad Hoc Conference
N
N
N
N
Y
Y
Meet-Me Conference
N
N
N
N
N
Y
Call Pickup
N
N
N
N
N
Y
Group Pickup
N
N
N
N
N
Y
Redial
N
N
N
N
Y7
Y7
Speed Dial
N
N
N
N
Y
Y
On-hook Dialing
N
N
N
N
N
N
Voice Mail Access
Y
Y
Y
Y
Y
Y8
Message Waiting Indicator (MWI)
N
N
N
N
Y
Y 8
Survivable Remote Site Telephony (SRST) Support
N
N
N
Y
Y
Y
Music on Hold (MoH)
Y
Y
Y
N
Y
Y
Mute
N
N
N
N
N
N
Multilevel Precedence and Preemption (MLPP)
N
N
N
N
N
N
Barge
N
N
N
N
N
N
cBarge
N
N
N
N
N
N
Call Preservation
N
N
N
N
Y9
N
Call Admission Control
Y
N
N
N
N
N
Local Voice Busy-Out
Y
N
N
N
N
N
Private Line Automatic Ringdown (PLAR)
Y
N
N
N
N
Y
Hunt Group
Y
N
N
N
N
N
Dial Plan Mapping
Y
N
N
N
N
N
Supervisory Disconnect
Y
N
N
N
N
N
Signaling Packet ToS Value Marking
0x68
0x6810
0x68
0x68
0x68
0x68
Media Packet ToS Value Marking
0xB8
0xB8
0xB8
0xB8
0xB8
0xB8
Fax Pass-Through
Y11
Y
Y12
Y
Y11
Y
Fax Relay
Y
Y
N
Y
Y
N
Skinny Client Control Protocol (SCCP)
N
N
N
N
Y
Y
Session Initiation Protocol (SIP)
N
N
N
Y
N
Y
H.323
Y
Y
N
Y
N
Y
Media Gateway Control Protocol (MGCP)
Y
Y
Y
Y
N
Y13
G.711
Y
Y
Y
Y
Y
Y
G.722
N
N
N
N
N
N
G.723
Y
Y
N
N
N
Y
G.726
Y
N
N
N
N
N
G.729
Y
Y
Y
Y
Y
Y
Voice Activity Detection (VAD)
Y
Y
N
Y
N
Y
Comfort Noise Generation (CNG)
Y
Y
N
Y
N
Y
1 Two 10/100 Base-T.
2 One 10/100 Base-T.
3 Two 10/100 Base-T for ATA 188; one 10 Base-T for ATA 186.
4 The EVM-HD-8FXS/DID provides eight ports on the baseboard and can be configured for FXS or DID signaling. In addition, it has room for two EM-HDA-8FXS as extension modules.
5 H.323 and SIP call control.
6 Call Forward All.
7 Last Number Redial.
8 Only on SCCP and SIP version.
9 Supported on VG248 version 1.2 or later.
10 It marks MGCP signaling on UDP port 2427, but it marks the MGCP keep-alive packets as best-effort on TCP port 2428.
11 Fax pass-through and fax relay.
12 Fax pass-through.
13 Cisco Unified CallManager does not support MGCP with the ATA.
Table 21-6 Cisco Basic IP Phones with SCCP
Ethernet Connection
Y1
Y1
Y2
Y1
Y3
Y3
Y3
Ethernet Switch (PC port)
N
N
Y
N
Y
Y
Y4
Cisco Power-Over-Ethernet (PoE)
Y
Y
Y
Y
Y
Y
Y
IEEE 802.3af Power-Over-Ethernet (PoE)
N
N
Y
N
N
Y
N
Localization
N
Y
Y
N
N
Y
Y
Directory Number
1
1
1
1
1
1
1
Maximum number of calls per line
200
200
200
200
200
200
200
Liquid Crystal Display
N
Y
Y
Y
Y
Y
Y
Caller ID
N
Y
Y
Y
Y
Y
Y
Call Waiting
N
Y
Y
Y
Y
Y
Y
Caller ID on Call Waiting
N
Y
Y
Y
Y
Y
Y
Call Hold
Y
Y
Y
Y
Y
Y
Y
Blind Transfer
N
N
N
N
N
N
N
Early-attended Transfer
Y
Y
Y
Y
Y
Y
Y
Consultative Transfer
Y
Y
Y
Y
Y
Y
Y
Call Forward
Y
Y
Y
Y
Y
Y
Y
Auto-Answer
N
Y5
Y5
N
N
Y5
Y5
Ad Hoc Conference
Y
Y
Y
Y
Y
Y
Y
Meet-Me Conference
N
Y
Y
Y
Y
Y
Y
Call Pickup
N
Y
Y
Y
Y
Y
Y
Group Pickup
N
Y
Y
Y
Y
Y
Y
Call Pickup Notification
N
Y
Y
N
N
Y
Y
Directed Call Park
N
N
N
N
N
N
N
Logging out of Hunt Groups
N
Y
Y
N
N
Y
Y
Redial
Y6
Y6
Y6
Y6
Y6
Y6
Y6
Speed Dial
Y
Y
Y
Y
Y
Y
Y
On-hook Dialing
N
Y
Y
Y
Y
Y
Y
Voice Mail Access
Y
Y
Y
Y
Y
Y
Y
Message Waiting Indicator (MWI)
Y
Y
Y
Y
Y
Y
Y
Video call
N
N
N
N
N
N
N
Survivable Remote Site Telephony (SRST) Support
Y
Y
Y
Y
Y
Y
Y
Unicast MoH
Y
Y
Y
Y
Y
Y
Y
Multicast MoH
Y
Y
Y
Y
Y
Y
Y
Tone on Hold
Y
Y
Y
Y
Y
Y
Y
Speaker
N
Y5
Y5
Y5
Y5
Y5
Y5
Headset Jack
N
N
N
N
N
N
N
Mute
N
N
N
Y
Y
N
N
Multilevel Precedence and Preemption (MLPP)
Y
Y
Y
Y
Y
Y
Y
Barge
N
N
N
N
N
N
Y
cBarge
N
Y
Y
N
N
Y
Y
Disable General Attribute Registration Protocol (GARP)
Y
Y
Y
Y
Y
Y
Y
Signaling and Media Encryption
N
N
N
N
N
N
N
Signaling Integrity
N
N
N
N
N
N
N
Manufacturing-Installed Certificate (X.509v3)
N
N
N
N
N
N
N
Field-Installed Certificate
N
N
N
N
N
N
N
Third-Party XML Service
N
Y
Y
N
N
Y
Y
External Microphone and Speaker
N
N
N
N
N
N
N
Dial plan
N
N
N
N
N
N
N
Signaling Packet ToS Value Marking
0x60
0x60
0x60
0x60
0x60
0x60
0x60
Media Packet ToS Value Marking
0xB8
0xB8
0xB8
0xB8
0xB8
0xB8
0xB8
G.711
Y
Y
Y
Y
Y
Y
Y
G.722
N
N
N
N
N
N
N
G.723
N
Y
N
N
N
N
N
G.726
N
Y
N
N
N
N
N
G.729
Y
Y
Y
Y
Y
Y
Y
Wideband Audio
N
N
N
N
N
N
N
Wideband Video
N
N
N
N
N
N
N
Voice Activity Detection (VAD)
Y
Y
Y
Y
Y
Y
Y
Comfort Noise Generation (CNG)
Y
Y
Y
Y
Y
Y
Y
Voice Quality Metrics
N
N
Y
N
N
Y
N
DTMF - SCCP
Y
Y
Y
Y
Y
Y
Y
DTMF - RFC2833
Y
Y
Y
Y
Y
Y
Y
1 One 10 Base-T.
2 One 10/100 Base-T.
3 Two 10/100 Base-T.
4 The Cisco Unified IP Phone 7912GA has an enhanced version of Ethernet switch.
5 One-way audio monitor mode.
6 Last Number Redial.
Table 21-7 Cisco Business, Manager, and Executive IP Phones with SCCP
Ethernet Connection
Y1
Y2
Y1
Y3
Y1
Y4
Ethernet Switch (PC port)
Y
Y
Y
Y
Y
Y
Cisco Power-Over-Ethernet (PoE)
Y
Y
Y
Y
Y5
Y
IEEE 802.3af Power-Over-Ethernet (PoE)
N
Y
N
Y
Y5
Y
Localization
Y
Y
Y
Y
Y
Y
Directory Number
2
2
6
6
8
8
Maximum number of calls per line
200
200
200
200
200
200
Liquid Crystal Display
Y
Y
Y
Y
Y
Y
Caller ID
Y
Y
Y
Y
Y
Y
Call Waiting
Y
Y
Y
Y
Y
Y
Caller ID on Call Waiting
Y
Y
Y
Y
Y
Y
Call Hold
Y
Y
Y
Y
Y
Y
Blind Transfer
N
N
N
N
N
N
Early-Attended Transfer
Y
Y
Y
Y
Y
Y
Consultative Transfer
Y
Y
Y
Y
Y
Y
Call Forward
Y
Y
Y
Y
Y
Y
Auto-Answer
Y
Y
Y
Y
Y
Y
Ad Hoc Conference
Y
Y
Y
Y
Y
Y
Meet-Me Conference
Y
Y
Y
Y
Y
Y
Call Pickup
Y
Y
Y
Y
Y
Y
Group Pickup
Y
Y
Y
Y
Y
Y
Call Pickup Notification
Y
Y
Y
Y
Y
Y
Directed Call Park
Y
Y
Y
Y
Y
Y
Logging out of Hunt Groups
Y
Y
Y
Y
Y
Y
Redial
Y6
Y6
Y6
Y6
Y6
Y6
Speed Dial
Y
Y
Y
Y
Y
Y
On-hook Dialing
Y
Y
Y
Y
Y
Y
Voice Mail Access
Y
Y
Y
Y
Y
Y
Message Waiting Indicator (MWI)
Y
Y
Y
Y
Y
Y
Video call
Y
Y
Y
Y
Y
Y
Survivable Remote Site Telephony (SRST) Support
Y
Y
Y
Y
Y
Y
Unicast MoH
Y
Y
Y
Y
Y
Y
Multicast MoH
Y
Y
Y
Y
Y
Y
Tone on Hold
Y
Y
Y
Y
Y
Y
Speaker
Y
Y
Y
Y
Y
Y
Headset Jack
Y
Y
Y
Y
Y
Y
Mute
Y
Y
Y
Y
Y
Y
Multilevel Precedence and Preemption (MLPP)
Y
Y
Y
Y
Y
Y
Barge
Y
Y
Y
Y
Y
Y
cBarge
Y
Y
Y
Y
Y
Y
Disable General Attribute Registration Protocol (GARP)
Y
Y
Y
Y
Y
Y
Signaling and Media Encryption
Y
Y
Y
Y
Y
Y
Signaling Integrity
Y
Y
Y
Y
Y
Y
Manufacturing-Installed Certificate (X.509v3)
N
Y
N
Y
Y
Y
Field-Installed Certificate
Y
N
Y
N
N
N
Third-Party XML Service
Y
Y
Y
Y
Y
Y
External Microphone and Speaker
Y
Y
Y
Y
Y
Y
Dial Plan
N
N
N
N
N
N
Signaling Packet ToS Value Marking
0x60
0x60
0x60
0x60
0x60
0x60
Media Packet ToS Value Marking
0xB8
0xB8
0xB8
0xB8
0xB8
0xB8
G.711
Y
Y
Y
Y
Y
Y
G.722
N
N
N
N
N
N
G.723
N
N
N
N
N
N
G.726
N
N
N
N
N
N
G.729
Y
Y
Y
Y
Y
Y
Wideband Audio
N
N
N
N
N
N
Wideband Video
N
N
N
N
N
N
Voice Activity Detection (VAD)
Y
Y
Y
Y
Y
Y
Comfort Noise Generation (CNG)
Y
Y
Y
Y
Y
Y
Voice Quality Metrics
Y
Y
Y
Y
Y
Y
DTMF - SCCP
Y
Y
Y
Y
Y
Y
DTMF - RFC2833
Y
Y
Y
Y
Y
Y
1 Two 10/100 Base-T.
2 The Cisco Unified IP Phone 7941G has two 10/100 Mbps Ethernet switches, and the Cisco Unified IP Phone 7941GE has two 10/100/1000 Mbps Ethernet switches.
3 The Cisco Unified IP Phone 7961G has two 10/100 Mbps Ethernet switches, and the Cisco Unified IP Phone 7961GE has two 10/100/1000 Mbps Ethernet switches.
4 The Cisco Unified IP Phone 7971G has two 10/100 Mbps Ethernet switches, and the Cisco Unified IP Phone 7971GE has two 10/100/1000 Mbps Ethernet switches.
5 The external power adaptor (CP-PWR-CUBE-3) must be used in order to have full display brightness.
6 Last Number Redial.
Table 21-8 Specialized Endpoints
Ethernet Connection
N
Y1
Y2
Ethernet Switch (PC port)
N
N
Y
Cisco Power-Over-Ethernet (PoE)
N
N
N
IEEE 802.3af Power-Over-Ethernet (PoE)
N
N
Y
Localization
Y
N
Y
Directory Number
12
1
2
Max number of calls per line
2
2
100
Liquid Crystal Display
Y
Y
Y
Caller ID
Y
Y
Y
Call Waiting
Y
Y
Y
Caller ID on Call Waiting
Y
Y
Y
Call Hold
Y
Y
Y
Blind Transfer
N
N
N
Early-Attended Transfer
Y
Y
Y
Consultative Transfer
Y
Y
Y
Call Forward
Y
Y
Y
Auto-Answer
N
N
Y
Ad Hoc Conference
Y
Y
Y
Meet-Me Conference
Y
Y
Y
Call Pickup
Y
Y
Y
Group Pickup
Y
Y
Y
Call Pickup Notification
N
N
N
Directed Call Park
N
N
N
Logging out of Hunt Groups
N
N
N
Redial
Y3
Y
Y
Speed Dial
Y
N
Y
On-hook Dialing
Y
Y
Y
Voice Mail Access
Y
N
Y
Message Waiting Indicator (MWI)
Y
N
Y
Video call
N
N
Y
Survivable Remote Site Telephony (SRST) Support
Y
Y
Y4
Unicast MoH
Y
Y
Y
Multicast MoH
Y
Y
N
Tone on Hold
Y
Y
Y
Speaker
N
Y
Y
Headset Jack
Y
N
Y
Mute
Y
Y
Y
Multilevel Precedence and Preemption (MLPP)
N
N
Y
Barge
N
N
Y
cBarge
N
N
Y
Disable General Attribute Registration Protocol (GARP)
N
N
N
Signaling and Media Encryption
Y
N
N
Signaling Integrity
N
N
N
Manufacturing-Installed Certificate (X.509v3)
N
N
N
Field-Installed Certificate
N
N
N
Third-Party XML Service
Y
N
N
External Microphone and Speaker
N
N
N
Signaling Packet ToS Value Marking
0x60
0x60
0x60
Media Packet ToS Value Marking
0xB8
0xB8
0x88
G.711
Y
Y
Y
G.722
N
N
Y
G.723
N
N
N
G.726
N
N
N
G.729
Y
Y
Y
Wideband Audio
N
N
N
Wideband Video
N
N
N
H.261
N
N
Y
H.263
N
N
Y
H.263+
N
N
Y
H.264
N
N
Y
Voice Activity Detection (VAD)
Y
Y
Y
Comfort Noise Generation (CNG)
Y
Y
Y
Voice Quality Metrics
N
N
N
DTMF - H.245
N
N
N
DTMF - SCCP
Y
Y
Y
DTMF - RFC2833
N
N
N
1 One 10/100 Base-T.
2 Two 10/100 Base-T.
3 Last Number Redial.
4 Only audio supported on SRST.
