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Collaboration Endpoints chapter of the Cisco Collaboration System 11.x SRND.
Cisco Collaboration System 1. Solution Reference Network Designs (SRND) - Collaboration Endpoints . Collaboration endpoints rely on enterprise call control and/or cloud- based collaboration for voice and video calling services. Collaboration endpoints also leverage both enterprise on- premises and cloud- based collaboration services such as voice messaging, instant messaging, and presence.
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Further, these endpoints gain key supplementary services from enterprise media resources such as video and voice conferencing, transcoding, and music on hold. Figure 8- 1 Cisco Collaboration Endpoints Architecture. While there are multiple options when deploying collaboration call control for voice and video services, each call control platform provides endpoint registration, call setup and routing services, and access to provisioned media resources. The high- level call control interactions between endpoints and the enterprise Cisco Unified Communications Manager is described in the following sub- section. Cisco Unified Communications Manager (Unified CM) Call Control. Call signaling in Cisco Unified Communications Manager (Unified CM), Cisco Business Edition, and Cisco Unified Communications Manager Express (Unified CME) distinguishes between line- side signaling and trunk- side signaling.
Whereas trunk- side signaling is used for connecting the entire call processing cluster or router to other servers and gateways, the line side is used for connecting endpoint devices to the call processing platform. The two interfaces are distinct in the services they offer, with the line side offering a rich set of user- oriented features. Session Initiation Protocol (SIP) and Skinny Client Control Protocol (SCCP) are the two main line- side signaling protocols supported by Cisco call processing platforms. All Cisco endpoints support either or both of these protocols. The set of features supported in both protocols is roughly equivalent, and the choice of which protocol to use is essentially a personal preference in a deployment. However, SIP is the protocol of choice for support of all new features and Cisco endpoints.
Cisco endpoints must be configured with several operating parameters before they can be used to make or receive calls or to run applications. This configuration must be performed in advance on the call processing server or router. Once configured, the call processing platform generates a configuration file for the endpoint to use, and it stores that file on a Trivial File Transfer Protocol (TFTP) server. The endpoints themselves go through a boot- up sequence when powered on. They retrieve this configuration file before they register with the appropriate server, and then they are ready to be used. The endpoints execute the following steps as part of the boot- up sequence: 1. When connected to the access switch, if the endpoint is not plugged in to a power source, it attempts to obtain power from the switch (Power over Ethernet).
Wireless and mobile endpoints are not connected to the enterprise network via Ethernet and therefore always derive power from a battery or power outlet. Once power is obtained, if device security is enabled, the endpoint presents its credentials to the security server or network authentication infrastructure. If it is allowed to use the network, the endpoint obtains its network parameters such as IP address, Domain Name Service (DNS) servers, gateway address, and so forth, either through static provisioning in the endpoint or through Dynamic Host Control Protocol (DHCP). The endpoint also obtains a TFTP server address either through static provisioning or through DHCP options. The endpoint then uses the TFTP server address to obtain its configuration files that, among other parameters, details the call processing server(s) or router(s) that the endpoint may associate and register with, the directory numbers that the endpoint must support, and so forth.
The endpoint registers with the call processing platform and is available for use. To confirm which endpoints support registration to Cisco Unified CM, refer to the endpoint data sheets listed in various other sections of this chapter.
Collaboration Endpoint Section 5. Conformance. Regardless of the call control platform you choose, when selecting endpoints and designing your Cisco Collaboration network you should strive to make the telephony features more accessible to users with disabilities, in conformance with Section 2. Telecommunications Act and U. S. Section 5. 08.
Observe the following basic design guidelines when configuring your Cisco Unified Communications network to conform to Section 5. Enable Quality of Service (Qo. S) and call admission control on the network to ensure optimal quality of voice and video so that enterprise communications are as clear and precise as possible. Configure only the G.
TTY) device or a Telephone Device for the Deaf (TDD). Although low bit- rate codecs such as G. TTY/TDD devices if they have an error rate higher than 1% Total Character Error Rate (TCER). Configure TTY/TDD devices for G.
WAN, if necessary. Enable (turn ON) Echo Cancellation for optimal performance. Voice Activity Detection (VAD) does not appear to have an effect on the quality of the TTY/TDD connection, so it may be disabled or enabled. However, Cisco recommends leaving VAD (also known as silence suppression) disabled on Unified CM call control and using the no vad command on H. Cisco IOS SIP dial- peers. Configure the appropriate regions and device pools in Unified CM to ensure that the TTY/TDD devices always use G. Awesome Tanks 2 Hacked Unlimited Ammo Resident. Connect the TTY/TDD to the Cisco Unified Communications network in either of the following ways: – Direct connection (Recommended method)Plug a TTY/TDD with an RJ- 1.
Cisco FXS port. Any Cisco voice gateway with an FXS port will work. Cisco recommends this method of connection.– Acoustic coupling. Place the IP phone handset into a coupling device on the TTY/TDD. Acoustic coupling is less reliable than an RJ- 1.
If stutter dial tone for audible message waiting indication (AMWI) is required, use an analog phone in conjunction with an FXS port on the Cisco VG Analog Gateways or Analog Telephony Adaptor (ATA). In addition, most Cisco IP Phones support stutter dial tone.
When you deploy immersive Cisco Tele. Presence rooms, ensure that ample room is provided to accommodate and provide for unimpeded movement of wheel chairs and other assistive conveyances. Analog Endpoints. An analog gateway typically is used to connect analog devices such as fax machines, modems, telecommunications device for the deaf (TDD)/teletypewriter (TTY), and analog phones, to the Vo. IP network so that the analog signal can be packetized and transmitted over the IP network. Analog gateways also provide physical connectivity to the PSTN and other traditional telephony equipment such as PBXs and key systems. Analog gateways include Cisco IOS router- based analog interface or service modules as well as fixed- port standalone gateways.
Generally analog gateways rely on Cisco Unified CM, Cisco Business Edition, Unified CM Express, and even Survivable Remote Site Telephony (SRST) for call control, supplementary services, and in some cases interface registration and configuration. Call control protocols supported across Cisco analog gateways include SIP, H. SCCP, and Media Gateway Control Protocol (MGCP). Standalone Analog Gateways. Cisco standalone analog gateways, including the Cisco Analog Telephony Adaptor (ATA) and Cisco VG Series Gateway, provide connectivity for analog devices such as fax machines, modems, TDD/TTY, paging systems, and analog phones, as well as one or more Ethernet ports for connecting to the IP network. Cisco standalone analog gateways support the FXS analog telephony interface port type only. For more information on Cisco ATAs, refer to the data sheets and documentation for the ATA 1.
Series at: http: //www. For more information on Cisco VG Series Gateways, refer to the data sheets and documentation at: http: //www. Analog Interface Module.