IP Telephony Endpoints

Major IP Telephony Endpoints (my quick notes)

» Analog Gateways
» Cisco Unified IP Phones
» Software-Based Endpoints
» Wireless Endpoints
» Cisco IP Conference Station
» Video Endpoints

Reference:

» IP Telephony Endpoints

Filed under: » Gatekeeper, » Gateway, » Voice

Review Notes: Gateway Deployments

A voice gateway allows terminals of one type, such as H.323, to communicate with terminals of another type, such as a PBX, by converting protocols.Gateways connect a company network to the PSTN, a PBX, or individual analog devices such as a phone or fax.

Types of Cisco access gateways:

• Analog Gateways:
  • Analog Station – gateways that connect an IP telephony network to plain old telephone service (POTS). They provide Foreign Exchange Station (FXS) ports to connect analog telephones, interactive voice response (IVR) systems, fax machines, PBX systems, and voice-mail systems.
  • Analog Trunk – gateways that connect an IP telephony network to the PSTN central office (CO) or a PBX. They provide Foreign Exchange Office (FXO) ports for PSTN or PBX access and recEive and transMit (E&M) ports for analog trunk connection to a legacy PBX. To minimize any answer and disconnect supervision issues, use digital gateways whenever possible. Analog direct inward dialing (DID) is also available for PSTN connectivity.

• Digital Gateways: Cisco access digital trunk gateways connect an IP telephony network to the PSTN or to a PBX via digital trunks, such as PRI common channel signaling (CCS), BRI, and T1 or E1 channel associated signaling (CAS). Digital T1 PRI trunks may also connect to certain legacy voice-mail systems.

IP telephony gateways should meet these core feature requirements:

• Gateway protocol support: Gateways support H.323, Media Gateway Control Protocol (MGCP), session initiation protocol (SIP), and Skinny Client Control Protocol (SCCP). H.323 and SIP gateways do not need a call control agent. MGCP and SCCP are streamlined protocols that only work on a network in which a call agent such as a Cisco Unified CallManager is present.

• Advanced gateway functionality:
  • Dual tone multifrequency (DTMF) relay capabilities: Each digit dialed with tone dialing is assigned a unique pair of frequencies. Voice compression of these tones with a low bit-rate codec can cause DTMF signal loss or distortion. Therefore, DTMF tones are separated from the voice bearer stream and sent as signaling indications through the gateway protocol (H.323, SCCP, or MGCP) signaling channel instead.
  • Supplementary services support: These services provide user functions such as hold, transfer, and conferencing, and are considered to be fundamental requirements of any voice installation.
  • Work with redundant Cisco Unified CallManagers: The gateways must support the ability to rehome to a secondary Cisco Unified CallManager in the event of a primary Cisco Unified CallManager failure.
  • Call survivability in Cisco Unified CallManager: The voice gateway preserves the Real-Time Transport Protocol (RTP) bearer stream (the voice conversation) between two IP endpoints when the Cisco Unified CallManager to which the endpoint is registered is no longer accessible.
  • Q Signaling (QSIG) support: QSIG is becoming the standard for PBX interoperability in Europe and North America. With QSIG, the Cisco voice packet network appears to PBXs as a distributed transit PBX that can establish calls to any PBX or other telephony endpoint served by a Cisco gateway, including non-QSIG endpoints. (For interoperability purposes)
  • Fax and modem support: Fax over IP enables interoperability of traditional analog fax machines with IP telephony networks. The fax image is converted from an analog signal and is transmitted as digital data over the packet network.

• Cisco Unified CallManager Release 3.1 and later supports H.323 and MGCP gateway protocols.
• Cisco Unified CallManager Release 4.0 and later also supports SIP.
• Cisco IP phones use SCCP, which is a lighter-weight protocol. SCCP uses a client/server model, while H.323 is a peer-to-peer model. MGCP also follows a client/server model.
• Protocol selection depends on site-specific requirements and the installed base of equipment. For example, most remote branch locations have Cisco 2600XM Series or 3700 Series multiservice routers installed. These routers support H.323 and MGCP 0.1 with Cisco IOS Release 12.2(11)T and Cisco Unified CallManager Release 3.1 or later. For gateway configuration, you might prefer MGCP to H.323 due to simpler configuration or, with older IOS versions, due to support for call survivability during a Cisco Unified CallManager failover from a primary to a secondary Cisco Unified CallManager. Failover with H.323 is only supported with Cisco Unified CallManager 4.1 or later and Cisco IOS Release 12.4(4)XC or later. On the other hand, you might prefer H.323 over MGCP because of the robustness of the interfaces supported.
• The Simplified Message Desk Interface (SMDI) is a standard for integrating voice mail systems to PBXs or Centrex systems. Connecting to a voice-mail system via SMDI and using either analog FXS or digital T1 PRI requires either SCCP or MGCP protocol because H.323 devices do not identify the specific line being used from a group of ports. The use of H.323 gateways for this purpose means the Cisco Messaging Interface cannot correctly correlate the SMDI information with the actual port or channel being used for an incoming call.

“He who wrestles with us strengthens our nerves and sharpens our skill. Our antagonist is our helper” – Edmund Burke, 1729-1797

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Filed under: » CCVP, » Gatekeeper, » Gateway, » Review Notes, » Voice