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Cisco AVVID and IP Telephony. Design & Implementation

Chapter 3. AVVID Gateway Selection

Paul J. Fong

Источник: Syngress Publishing Inc, 2001

Introduction

Gateways are part of the network platform’s components, which is a sublayer of the network infrastructure layer of AVVID (Architecture for Voice,Video, and Data). Moving to an AVVID architecture requires integrating with an existing PBX (Private Branch eXchange) and PSTN (Public Switched Telephone Network) infrastructure. Whether there are existing analog or digital voice cir­cuits in place, gateways need to be implemented to merge the legacy architecture to an IP-based voice, video, and data network. In addition, voice gateways provide connectivity from the new AVVID infrastructure to legacy voice mail systems.
Gateway selection is a topic often overlooked. When designing an AVVID network, careful consideration should be given to the type of gateway you need to implement within an AVVID network; certain platforms support particular gateway protocols.

Introduction to AVVID Gateways

A gateway, by definition, is a device that converts one protocol to another. In the AVVID or Voice over IP (VoIP) environment, a gateway is respon­sible for connecting an IP telephone network to the PSTN or PBX and key sys­tems. For example, the gateway may connect an H.323 network to an SIP-based network, PSTN, or ISDN. It also performs translations between different trans­mission formats and communication procedures, and is responsible for setting up and clearing calls on both sides. Communication between terminals and gateways is done through the H.245 and Q.931 protocols.
Based on the device or the implementation, the gateways communicate with Cisco CallManager or other network devices over various gateway protocols.Your own infrastructure and VoIP requirements will help determine what gateway is right for you, but required common features include: DTMF, CallManager redundancy, and supplementary services. Supplementary services allow users to perform call hold, transfer, and conferencing.

Understanding the Capabilities of Gateway Protocols

The three voice gateway protocols supported in Cisco’s AVVID architecture are Skinny Station Protocol (SSP), H.323, and MGCP (Media Gateway Control Protocol). Skinny Station Protocol allows a Skinny client to use TCP/IP to transmit and receive calls and RTP/UDP/IP packets for audio. An example of a Skinny client is an IP phone or gateway. The Skinny clients communicate with a Cisco CallManager over TCP on ports 2000–2002. SSP was developed by Cisco as a low-bandwidth gateway protocol.
H.323 is the most supported gateway protocol from Cisco, and is an ITU-T (Telecommunication Standardization Sector of the International Telecommunica­tions Union) standard for packet-based audio, video, and conferencing. It is the standard for the conferencing standard (made up of others such as H.245, H.225, and Q.931), and is the only gateway that provides full routing capabilities. It transmits and receives media streams via RTP with Real-Time Control Protocol (RTCP), carried over UDP, thereby providing status and control information. Q.931 signaling is for call setup and termination. Capabilities, however, are exchanged by utilizing H.245, which is for call control, and establishes multimedia communication or call services between the H.323 clients.
The MGCP protocol functions in an architecture where the call control intelligence is removed from the gateway. Level3, Bellcore, Cisco, and Nortel developed MGCP which is a master/slave protocol, where the gateway is the slave servicing commands from the master, which is the call agent. In the Cisco AVVID environment, the CallManager functions as the call agent.
The skinny gateways are the DT-24, DE-30, and Catalyst 4000/6000 modules, which provide CallManager access to digital gateways. An example of an H.323 gateway is a Cisco IOS router like the 2600 and 3600. The VG-200 is an MGCP gateway with future support for the 2600, 3600, 3810, and Catalyst modules.
Another protocol being implemented in Cisco gateways is the Session Initiation Protocol. SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions or calls. These sessions include IP conferences, telephone calls, and multimedia distribu­tion. A Cisco VoIP solution for SIP consists of a SIP agent, 7960 IP Phone, SIP gateway, and a SIP proxy server.
SIP supports five elements of establishing and terminating communications:

1. User location
2. User capabilities
3. User availability
4. Call setup
5. Call handling

Currently, the VoIP world is dominated by H.323; the emergence of SIP and the increasing number of applications supporting this new technology means the interoperability of SIP with existing H.323 networks.
 

Note

An example of new software utilizing the functionality of SIP is the appli­cation Windows Messenger, which is part of Windows XP. Windows Messenger is real-time communications software that provides end-to-end IP telephony.
Video gateways are used to convert form H.320 devices to H.323 devices. The Cisco IP/VC products allow companies to utilize their legacy H.320 ISDN-based videoconferencing to integrate with newer IP-based H.323 videoconfer­encing devices. Cisco IP/VC gateways support H.261 and H.263 video coders/ decoders ; H.261 is used as multiple channels of 64 Kbps, while H.263 is a higher quality video

Table 3.1 Video Image Format Standards

Choosing a Voice Gateway Solution

There are a number of different voice gateways available for CallManager and VoIP implementations, which are divided into categories by type of gateway and the protocol. The gateway selection is based on some of the following variables: analog or digital, capacity, connection type, services, features. Table 3.2 lists the analog VoIP gate­ways and the respective voice interface cards (VICs) supported. The analog gate­ways provide connectivity to analog phone sets, central office, and PBX. The Foreign Exchange Station (FXS) ports are used to provide dial tone for analog phones, faxes, while a Foreign Exchange Office (FXO) port in a gateway is for connectivity to Central Office for analog access to the PSTN. Ear-and-mouth (E&M) ports, on the other hand, are for PBX-to-PBX signaling communication. You can determine what type of analog VoIP gateway you need by answering one of the following questions: Are FXO ports required for PSTN connectivity or is Direct Inward Dial (DID) a necessity? Another factor in selecting an analog gateway may be capacity. For example, if you require a large number of FXS ports for legacy analog connections, you would select a Cisco 3660 or a Catalyst 6000 with a 24-port FXS module rather than a smaller capacity gateway such as a VG-200 or Cisco 2600.

Table 3.2 Analog VoIP Gateways

If higher capacity voice channels are required to either the PSTN or PBX, a digital gateway may be more effective. Table 3.3 lists the interfaces and features supported on the various hardware platforms. The different gateways support two main signaling types: either ISDN Primary Rate Interface (PRI) or channel asso­ciated signaling (CAS) for T1 or E1. ISDN PRI, meanwhile, utilizes a “D” channel for signaling. ISDN PRI is classified as out-of-band signaling since there is a channel dedicated for signaling, whereas, CAS signaling uses some of the bandwidth from each channel. .T1CAS supports auto­matic number identifier (ANI) and dialed number identification service (DNIS) as well, which are also known as Caller ID and Called Party Number.  Determining which PRI type of inter­face is required depends on whether you’re connecting your gateway to a PBX or PSTN. Typically, if the gateway is connecting to a PBX, you will need a Network Side PRI interface, since the PBX is on the “user side.” Normally, the PSTN (with a switch like a DMS100) functions as the “network side” and the gateway needs a User Side PRI interface.

In choosing a gateway, attention should be given to ensure it supports three important features:

1. CallManager Redundancy
2. DTMF Relay
3. Supplemental Services

CallManager Redundancy is required since an AVVID network needs to have the same high level of avail­ability as the traditional PBX. DTMF uses two frequencies, a high and low tone to distinguish numbers on a telephone keypad. This signaling is usually carried over a 64 Kbps voice circuit, and accomplished with little problem, but with lower-bit CODEC the signal can be lost or unrecognizable. The gateways provide out-of-band support for passing DTMF signals across the VoIP network via the gateway protocols. The AVVID gateway needs to provide support for other user telephony services such as Call Hold, Call Handling, and Conference. These are normal traditional voice services, which should be considered basic requirements for an AVVID network.

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