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Cisco Unified Communications SRND Based on Cisco Unified Communications Manager 6.x
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Preface
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Introduction
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Unified Communications Deployment Models
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Network Infrastructure
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Gateways
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Cisco Unified CM 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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Unified Communications Endpoints
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Device Mobility
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Cisco Unified Presence
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Cisco Unified CM Applications
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Cisco Mobility Applications
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Network Management
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Cisco Unified Communications Manager Business Edition
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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
Unified Communications Endpoints
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 7910G and 7910G+SW
Cisco Unified IP Phone 7941G-GE
Cisco Unified IP Phone 7961G-GE
Cisco Unified IP Phone 7971G-GE
Cisco Unified IP Phone Expansion Module 7914
Cisco Unified Personal Communicator
Cisco Unified IP Conference Station
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 Phones
Cisco Unified Video Advantage with a Cisco Unified IP Phone
Sony and Tandberg SCCP Endpoints
Unified Communications Endpoints
This chapter summarizes various types of Unified Communications endpoints along with their features and QoS recommendations. The endpoints can be categorized into the following major types:
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Cisco Unified IP Conference Station
The sections listed above 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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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 an Skinny Client Control Protocol (SCCP) or Media Gateway Control Protocol (MGCP) endpoint with Cisco Unified Communications Manager (Unified CM) and can re-home to a Survivable Remote Site Telephone (SRST) router in failover scenarios. The Cisco VG224 supports Cisco Unified CM Release 3.1 and later. The Cisco VG224 also supports modem pass-through, modem relay, fax pass-through, and fax relay. In addition, the Cisco VG224 can be used to connect analog phones for SCCP support on Cisco Unified Communications Manager Express (Unified CME) and Cisco Unified Survivable Remote Site Telephone (SRST).
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 CM (Release 3.1 and later) and voice network. The Cisco VG248 also supports Unified CM 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. The Cisco ATA 186 and 188 can be configured in any of the following ways:
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The SCCP-based ATA behaves like an SCCP IP phone. The Cisco ATA 186 or 188 can be configured as a SIP client that registers with the SIP proxy server to make phone calls with another endpoint. The Cisco ATA 186 or 188 can act as either a user agent client (UAC) when it initiates SIP requests or as a user agent server (UAS) when it responds to requests. Cisco Unified CM 5.x does not have native SIP support for the Cisco ATA 186 or 188.
Cisco Unified IP Phones
The Cisco IP phone portfolio includes basic IP phones, business IP phones, manager IP Phones, and executive IP phones.
Cisco Basic IP Phones
The Cisco basic IP phone is best suited for low-traffic users with limited call features and budget requirements. The basic IP phones include Cisco Unified IP Phone 7902G, 7905G, 7906G, 7910G, 7910G+SW, 7911G, and 7912G.
Cisco Unified IP Phone 7902G
The Cisco Unified IP Phone 7902G supports a single line and it has a single 10-Base-T Ethernet port on the back of the phone. The Cisco Unified IP Phone 7902G does not have any liquid crystal display (LCD) screen. The Cisco Unified IP Phone 7902G supports SCCP but has no support for SIP.
Cisco Unified IP Phone 7905G
The Cisco Unified IP Phone 7905G supports a single line and it has a single 10-Base-T Ethernet port on the back of the phone. The speaker operates in one-way listen mode only. The Cisco Unified IP Phone 7905G supports SCCP and SIP; however, the features and user interface (UI) are not consistent across the two call signaling protocols.
Cisco Unified IP Phone 7906G
The Cisco Unified IP Phone 7906G supports a single line and it has a single 10/100 Base-T Ethernet port on the back of the phone. The speaker operates in one-way listen mode only. Power is supplied via IEEE 802.3af, Cisco inline power, or local power through a power adaptor (CP-PWR-CUBE-3). The Cisco Unified IP Phone 7906G supports SCCP and SIP and is one of the phones included in the enhanced architecture of Cisco Desktop IP Phones. This architecture provides for feature and UI consistency across the Cisco Desktop IP Phones regardless of call signaling protocol. The end-user experience for a supported feature will behave consistently whether using SCCP or SIP call control signaling.
Cisco Unified IP Phone 7910G and 7910G+SW
The Cisco Unified IP Phone 7910G supports only a single line, and the speaker operates in one-way listen mode only. The Cisco Unified IP Phone 7910G also has six feature access keys that can be configured in the customized phone button template by the administrator to provide the end-user with various call features. Because there are only six feature access keys on this phone model, one phone button template cannot provide the end-user with all the call features. Both the Cisco Unified IP Phone 7910G and 7910+SW support SCCP but have no support for SIP. The only difference between the Cisco Unified IP Phone 7910G and 7910G+SW is that the former has a 10 Base-T Ethernet port and the latter has two 10/100 Base-T Ethernet ports.
Cisco Unified IP Phone 7911G
The Cisco Unified IP Phone 7911G supports only a single line and it has two 10/100 Base-T Ethernet connections. The speaker operates in one-way listen mode only. The Cisco Unified IP Phone 7911G supports SCCP and SIP and is one of the phones included in the enhanced architecture of Cisco Desktop IP Phones. This architecture provides for feature and UI consistency across the Cisco Desktop IP Phones regardless of call signaling protocol. The end-user experience for a supported feature will behave consistently whether using SCCP or SIP call control signaling.
Cisco Unified IP Phone 7912G
The Cisco Unified IP Phone 7912G supports only a single line and it has two 10/100 Base-T Ethernet connections. The speaker operates in one-way listen mode only. The Cisco Unified IP Phone 7912G supports SCCP and SIP; however, the features and user interface (UI) are not consistent across the two call signaling protocols.
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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 Phones 7931G, 7940G, 7941G, 7941G-GE, 7942G, and 7945G.
Cisco Unified IP Phone 7931G
The Cisco Unified IP Phone 7931G supports up to 24 directory numbers that can be assigned to 24 illuminated line keys, and it is most suitable for retail, commercial, and manufacturing users. The Cisco Unified IP Phone 7931G has two 10/100 Base-T Ethernet connections and it supports SCCP only. In addition to the programmable softkey support available on other Cisco Unified IP Phones, the Cisco Unified IP Phone 7931G also has three dedicated keys for Hold, Redial, and Transfer features. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7940G
The Cisco Unified IP Phone 7940G can have up to two directory numbers and includes two 10/100 Base-T Ethernet connections. The Cisco Unified IP Phone 7940G supports SCCP and SIP; however, the features and user interface (UI) are not consistent across the two call signaling protocols. For example, the Cisco Unified IP Phone 7940G using SCCP has full security capability, whereas SIP does not have any previously implemented security features. The Cisco Unified IP Phone 7940G using SCCP is compatible with the Cisco Unified Video Advantage video-enabled endpoint to make video calls, whereas the Cisco Unified IP Phone 7940G using SIP has no video support. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7941G
The Cisco Unified IP Phone 7941G can have up to two directory numbers and includes two 10/100 Base-T Ethernet connections. The Cisco Unified IP Phone 7941G supports SCCP and SIP and is included in the enhanced architecture of Cisco Unified IP Phones. This architecture provides for feature and UI consistency across the Cisco IP phones regardless of call signaling protocol. The end-user experience for a supported feature will behave consistently whether using SCCP or SIP call control signaling.
There are a few features that are not supported with SIP that are supported with SCCP. For example, the Cisco Unified IP Phone 7941G using SCCP is compatible with the Cisco Unified Video Advantage video-enabled endpoint to make video calls, whereas SIP has no video support. The Cisco Unified IP Phone 7941G using SCCP supports Tone on Hold, whereas SIP has no Tone on Hold support. This phone includes a higher-resolution, 4-bit grayscale display for enhancing feature usage and Extensible Markup Language (XML) applications, as well as for enabling support for double-byte languages. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7941G-GE
The Cisco Unified IP Phone 7941G-GE can have up to two directory numbers and is the equivalent of the Cisco Unified IP Phone 7941G with the exception that it includes two 10/100/1000 Base-T Ethernet connections. The addition of gigabit throughput capability allows for high bit-rate and bandwidth-intensive applications on a co-located PC.
Cisco Unified IP Phone 7942G
The Cisco Unified IP Phone 7942G, like the 7941G, can have up to two directory numbers and includes two 10/100 Base-T Ethernet connections. In addition to the 7941G's other features and support for protocols, the 7942G adds support for the G.722 wideband codec and updates the speaker, microphone, and handset for high-fidelity voice communications. For a complete list of supported features, see the Endpoint Features Summaryy.
Cisco Unified IP Phone 7945G
The Cisco Unified IP Phone 7945G extends the features of the 7942G. Like the 7942G, the 7945G can have up to two directory numbers; but unlike the 7942G, the 7945G also includes two 10/100/1000 Base-T Ethernet connections and a five-way navigation button set. In addition to the support for G.722 wideband codec and high-fidelity speaker, microphone, and handset, the 7945G adds a backlit TFT color display for easy access to communications information, timesaving applications, and feature usage. For a complete list of supported features, see the Endpoint Features Summary.
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, 7961G-GE, 7962G, and 7965G.
Cisco Unified IP Phone 7960G
The Cisco Unified IP Phone 7960G can have up to six directory numbers and includes two 10/100 Base-T Ethernet connections. The Cisco Unified IP Phone 7960G supports SCCP and SIP; however, the features and user interface (UI) are not consistent across the two call signaling protocols. For example, the Cisco Unified IP Phone 7960G using SCCP has full security capability, whereas SIP does not have any previously implemented security features. The Cisco Unified IP Phone 7960G using SCCP is compatible with the Cisco Unified Video Advantage video-enabled endpoint to make video calls, whereas the Cisco Unified IP Phone 7960G using SIP has no video support. The Cisco Unified IP Phone 7960G using SCCP supports the Cisco Unified IP Phone Expansion Module 7914, whereas SIP has no support for the expansion module. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7961G
The Cisco Unified IP Phone 7961G can have up to six directory numbers and includes two 10/100 Base-T Ethernet connections. The Cisco Unified IP Phone 7961G supports SCCP and SIP and is included in the enhanced architecture of the Cisco Unified IP Phones. This architecture provides for feature and UI consistency across the Cisco IP phones regardless of call signaling protocol. The end-user experience for a supported feature will behave consistently whether using SCCP or SIP call control signaling.
There are a few features that are not supported with SIP that are supported with SCCP. For example, the Cisco Unified IP Phone 7961G using SCCP is compatible with the Cisco Unified Video Advantage video-enabled endpoint to make video calls, whereas SIP has no video support. The Cisco Unified IP Phone 7961G using SCCP supports Tone on Hold, whereas SIP has no Tone on Hold support. The Cisco Unified IP Phone 7961G using SCCP supports the Cisco Unified IP Phone Expansion Module 7914, whereas SIP has no support for the expansion module. This phone includes a higher-resolution, 4-bit grayscale display for enhancing feature usage and Extensible Markup Language (XML) applications, as well as for enabling support for double-byte languages. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7961G-GE
The Cisco Unified IP Phone 7961G-GE can have up to six directory numbers and is the equivalent of the Cisco Unified IP Phone 7961G with the exception that it includes two 10/100/1000 Base-T Ethernet connections. The addition of gigabit throughput capability allows for high bit-rate and bandwidth-intensive applications on a co-located PC.
Cisco Unified IP Phone 7962G
The Cisco Unified IP Phone 7962G, like the 7961G, can have up to six directory numbers and includes two 10/100 Base-T Ethernet connections. In addition to the 7961G's other features and support for protocols, the 7962G adds support for the G.722 wideband codec and updates the speaker, microphone, and handset for high-fidelity voice communications. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7965G
The Cisco Unified IP Phone 7965G extends the features of the 7962G. Like the 7962G, the 7965G can have up to six directory numbers; but unlike the 7962G, the 7965G also includes two 10/100/1000 Base-T Ethernet connections and a five-way navigation button set. In addition to the support for G.722 wideband codec and high-fidelity speaker, microphone, and handset, the 7965G adds a backlit TFT color display for easy access to communications information, timesaving applications, and feature usage. For a complete list of supported features, see the Endpoint Features Summary.
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, 7971G-GE, and 7975G.
Cisco Unified IP Phone 7970G
The Cisco Unified IP Phone 7970G can have up to eight directory numbers, has a high-resolution color touch screen, and has more access keys on the phone compared to other Cisco Unified IP Phones. The Cisco Unified IP Phone 7970G supports both SCCP and SIP and is included in the enhanced architecture of Cisco Desktop IP Phones. This architecture provides for feature and UI consistency across the Cisco Desktop IP Phones regardless of call signaling protocol. The end-user experience for a supported feature will behave consistently whether using SCCP or SIP call control signaling.
There are a few features that are not supported with SIP that are supported with SCCP. For example, the Cisco Unified IP Phone 7970G using SCCP is compatible with the Cisco Unified Video Advantage video-enabled endpoint to make video calls, whereas SIP has no video support. The Cisco Unified IP Phone 7970G using SCCP supports Tone on Hold, whereas SIP has no Tone on Hold support. The Cisco Unified IP Phone 7970G using SCCP supports the Cisco Unified IP Phone Expansion Module 7914, whereas SIP has no support for the expansion module. This phone includes a higher-resolution, 4-bit grayscale display for enhancing feature usage and Extensible Markup Language (XML) applications, as well as for enabling support for double-byte languages. For a complete list of supported features, see the Endpoint Features Summary.
Cisco Unified IP Phone 7971G-GE
The Cisco Unified IP Phone 7971G-GE can have up to eight directory numbers and is the equivalent of the Cisco Unified IP Phone 7970G with the exception that it includes two 10/100/1000 Base-T Ethernet connections. The addition of gigabit throughput capability allows for high bit-rate and bandwidth-intensive applications on a co-located PC.
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Cisco Unified IP Phone 7975G
The Cisco Unified IP Phone 7975G, like the 7971G-GE, can have up to eight directory numbers and two 10/100/1000 Base-T Ethernet connections. Unlike the 7971G-GE, however, the 7975G adds the G.722 wideband codec and high-fidelity speaker, microphone, and handset. The 7975G also has a touchscreen color display. For a complete list of supported features, see the Endpoint Features Summary.
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 Unified Personal Communicator and Cisco IP Communicator. A software-based endpoint is an application installed on a client PC, and it registers with (and is controlled by) Unified CM.
Cisco Unified Personal Communicator
Cisco Unified Personal Communicator is a software application based on Microsoft Windows or Macintosh. Cisco Unified Personal Communicator integrates a wide variety of communications applications and services into a single desktop application to help people communicate effectively. It allows users to access a variety of powerful communications tools, including voice, video, call management, presence, and web conferencing. The integrated applications include Cisco Unified Communications Manager (Unified CM), Cisco Unified Presence, Cisco Unity, Cisco Unity Connection, Cisco Unified MeetingPlace, Cisco Unified MeetingPlace Express, Cisco Unified Videoconferencing and MeetingPlace Express VT, and the Lightweight Directory Access Protocol (LDAP) version 3 (v3) server. For more information on Cisco Unified Personal Communicator, see the chapter on Cisco Unified Presence, page 23-1.
Regardless of the device limits allowed per server, there are maximum limits on the number CTI devices you can configure in Unified CM. The CTI device limits as they apply to Cisco Unified Personal Communicator are as follows:
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The following assumptions apply to the preceding maximum Cisco Unified Personal Communicator limits:
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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.
Because Cisco IP Communicator is a standalone device that supports both SCCP and SIP, 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 Unified Communications Deployment Models, page 2-1, for details.
The end-user experience for a supported feature is the same whether using SCCP or SIP call control signaling. There are a few features that are not supported with SIP that are supported with SCCP. For example, Cisco IP Communicator using SCCP is compatible with the Cisco Unified Video Advantage video-enabled endpoint for making video calls, whereas SIP has no video support. In addition, Cisco IP Communicator using SCCP supports Tone on Hold, whereas SIP has no Tone on Hold support. For a complete list of supported features, see the Endpoint Features Summary.
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 an area where traditional wired phones are undesirable or problematic. (Refer to Wireless LAN Infrastructure, page 3-62, for more information about wireless network design.)
Cisco offers the following two Voice over WLAN (VoWLAN) IP phones:
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Both are hardware-based phones with built-in radio antenna. The Cisco Unified Wireless IP Phone 7920 enables 802.11b wireless LAN connectivity to the network, while the Cisco Unified Wireless IP Phone 7921G enables 802.11b, 802.11g, or 802.11a connectivity to the network. These phones register with Unified CM using Skinny Client Control Protocol (SCCP), just like the hardware-based phones and Cisco IP Communicator. For more information about these phones, refer to the appropriate phone documentation available at
Site Survey
Before deploying the Cisco Unified Wireless IP Phones, 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 Phones (accessed via Menu > Network Config > Site Survey on the 7920 and via Settings > Status > Site Survey on the 7921G) 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 and 7921G 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 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 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 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 is the 802.11i enhanced version of WPA, which uses Advanced Encryption Standards (AES) rather than TKIP for encrypting traffic to and from the wireless device. This method is supported only on the Cisco Unified Wireless IP Phone 7921G.
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This method allows the Cisco Unified Wireless IP Phone to be authenticated to the AP via the configuration of a shared key on the Cisco Unified Wireless IP Phone 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 is the 802.11i enhanced version of WPA-PSK, which uses AES rather than TKIP for encrypting traffic to and from the wireless device. This method is supported only on the Cisco Unified Wireless IP Phone 7921G.
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This method allows the Cisco Unified Wireless IP Phone 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 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 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 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 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
The capacity of each AP depends on a number of factors including the AP radio types, associated client radio types, enabled data rates, and channel utilization.
Given an 802.11b-only AP with 802.11b clients and a data rate of 11 Mbps, the 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.
When using 802.11a at a data rate of 54 Mbps, the maximum number of active voice streams increases to between 14 and 18 per AP.
For 802.11g environments with a data rate of 54 Mbps, in theory the maximum number of active voice streams also increases to between 14 and 18 per AP. However, because most 802.11g environments are mixed and include 802.11b clients (and therefore 11 Mbps data rates) as well as 802.11g clients, capacity is typically significantly lower with a maximum of 8 to 12 active voice streams per AP.
Regardless of 802.11 radio type, call capacity can be diminished significantly if there is heavy channel utilization due to data traffic.
For additional information about call capacity, radio types, and data rates, refer to the VoWLAN design recommendations in the latest version of the Enterprise Mobility Design Guide, available at
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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 Phones that each AP can support. For example, given an 802.11b AP with 802.11b clients and a typical user-to-call capacity ratio of 3:1, a single AP can support 21 to 24 Cisco Unified Wireless IP Phones, depending on whether the codec used is G.711 or G.729. As another example, given an 802.11a AP with an 802.11a client at a data rate of 54 Mbps and user-to-call capacity ratio of 3:1, a single AP can support 42 to 54 Cisco Unified Wireless IP Phone 7921Gs. However, these numbers do not take into consideration the possibility that other Cisco Unified Wireless IP Phones could roam to the AP, so a lower number of phones per AP might be more realistic.
These capacities are based on voice activity detection (VAD) being 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 Unified CM. (It is referred to as Silence Suppression in Unified CM.) Thus, if VAD is enabled for the Cisco Unified Wireless IP Phone 7920, then it will be enabled for all devices in the Unified CM 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 3-47.
Phone Configuration
For information on configuring the Cisco Unified Wireless IP Phones, refer to the Cisco Unified Wireless IP Phone 7920 Administration Guide and the Cisco Unified Wireless IP Phone 7921G Administration Guide, available at
Roaming
Cisco Unified Wireless IP Phones are 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 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.
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WMM TSPEC is an 802.11e QoS mechanism that assists wireless IP phone roaming by enabling the phone to request bandwidth and priority treatment via a TSPEC indication while roaming to determine if the new AP can handle the phones bandwidth needs based on current utilization. TSPEC is supported only by the Cisco Unified Wireless IP Phone 7921G.
When devices roam at Layer 3, they move from one AP to another AP across native VLAN boundaries. When the WLAN network infrastructure consists of autonomous APs, the Cisco Catalyst 6500 Series Wireless LAN Services Module (WLSM) allows the Cisco Unified Wireless IP Phone to keep its IP address and roam at Layer 3 while still maintaining an active call. 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 Cisco WLSM product documentation available at
Seamless Layer 3 roaming for clients across a lightweight access point infrastructure is accomplished by WLAN controllers that use dynamic interface tunneling. Cisco Unified Wireless IP Phones that roam across WLAN controllers and VLANS can keep their IP address when using the same SSID and therefore can maintain an active call.
With stronger authentication methods such as WPA and EAP, the number of information exchanges increases and causes more delay during roaming. To avoid additional delays, use Cisco Centralized Key Management (Cisco CKM) to manage authentication. With Cisco CKM, whether at Layer 2 or Layer 3, roaming can occur without any perceptible delay. Cisco CKM also takes some of the load off the Access Control Server (ACS) by reducing the number of authentication requests that must be sent to the ACS.
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AP Call Admission Control
Call admission control mechanisms in Unified CM or in a gatekeeper can control WAN bandwidth utilization and provide QoS for existing calls, but both mechanisms are applied 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 as Cisco Unified Wireless IP Phones, there must also be a call admission control mechanism at the AP level because these wireless devices may roam from one AP to another.
Cisco APs and wireless voice clients have two mechanisms that are used for call admission control:
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QBSS 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 is not a true call admission control mechanism because it does not guarantee that calls will retain proper QoS or that there is enough bandwidth to handle the call. When a Cisco Unified Wireless IP Phone 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 oversubscription of the available bandwidth on that AP.
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WMM TSPEC is the QoS mechanism that enables WLAN clients to provide an indication of their bandwidth and QoS requirements so that APs can react to those requirements. When a client is preparing to make a call, it sends an Add Traffic Stream (ADDTS) message to the AP with which it is associated, indicating TSPEC. The AP can then accept or reject the ADDTS request based on whether bandwidth and priority treatment are available. If the call is rejected, the phone will receive a Network Busy message. When roaming, mid-call clients supporting TSPEC will send a ADDTS message to the new AP as part of the association process to ensure that there is available bandwidth for priority treatment. If there is not enough bandwidth, the roam can be load-balanced to a neighboring AP if one is available.
Cisco Unified Wireless IP Phone 7920s support only QBSS, so this is the only mechanism that can be used for call admission control with these devices. However, the Cisco Unified Wireless IP Phone 7921Gs support both QBSS and TSPEC. (TSPEC takes precedence over QBSS.) Therefore call admission control with the Cisco Unified Wireless IP Phone 7921G, when using TSPEC, is more accurate and eliminates the possibility of priority bandwidth oversubscription on the AP.
Cisco Unified IP Conference Station
The Cisco Unified IP Conference Station combines conference room speaker-phone technology with Cisco Unified Communications technology. The Cisco Unified 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 Unified CM 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.
The Cisco Unified IP Conference Station 7937G adds wideband acoustics, expanded room coverage, a larger backlit LCD, and an extra softkey. The Cisco Unified IP Conference Station 7937G also supports IEEE 802.3af Power over Ethernet, or it can also use the Cisco Power Cube. The Cisco Unified IP Conference Station 7937G requires Cisco Unified CM Release 4.1 or later.
Video Endpoints
Cisco Unified CM 5.x supports the following types of video-enabled endpoints:
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SCCP Video Endpoints
SCCP video endpoints register directly with Unified CM 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 7911, 7940, 7941, 7942, 7945, 7960, 7961, 7962, 7965, 7970, 7971, or 7975 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 SCCP 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 Unified CM 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 associate with it. (See Figure 21-1.)
Cisco Unified Video Advantage 2.0 does not rely on CDP to discover the presence of Cisco IP Communicator running SCCP on the same PC. Instead, it listens for a private Windows message sent from the Cisco IP Communicator process. If Cisco IP Communicator is discovered, the association process works exactly as it does for a hardware IP phone. (See Figure 21-2.)
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Figure 21-1 Cisco Unified Video Advantage Operational Overview
Figure 21-1 illustrates the following events:
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Figure 21-2 Cisco IP Communicator Associating with Cisco Unified Video Advantage
Figure 21-2 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.
When using a hardware IP Phone, the 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. When using Cisco IP Communicator, 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. Unified CM 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 lower 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 codec supports 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/en/US/products/sw/voicesw/ps5662/products_data_sheet0900aecd8044de04.html
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 Base-T 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 products that support the Cisco Skinny Client Control Protocol (SCCP). 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 Unified CM 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
