IP Telephone Design Basics

All manufacturers base their IP telephone on proprietary design schematics and circuitry, but there are common design elements across the unique terminals. IP telephone basics include:

• User interface

• Voice interface

• Network interface

• Processor complex and associated logic

The accompanying diagram illustrates the internal design elements of an IP telephone instrument (Figure 1)

Figure 1: IP telephone design.

The user interface elements provide four classic telephone user function interfaces: keypad for dialing numbers; a variety of keys for line and feature access; a display for user prompts, caller feedback, messages, and other call processing information; serial interface to allow communications to an external device, such as a PDA, to allow synchronization of telephone information; speed dialing; and customer programming. An audible indicator (ringer) is also included to announce incoming calls.

The voice interface converts input analog voice signals into 8-bit digital word bit samples. Speech signals are sampled at an 8-KHz rate to create a 64-Kbps digital bit stream to the processor by using a standard PCM codec. Voice signal compression and IP encoding functions are performed by processor complex elements. The processor complex performs voice processing, call processing, protocol processing, and network management software functions. The complex consists of a DSP for voice-related functions and a MCU for the remaining control and management functions. The DSP and MCU each have associated memory. DSP memory usually includes RAM and ROM elements; MCU memory usually includes RAM and Flash elements. The Flash memory element supports software upgrades.

The network interface allows the transmission and reception of voice packets to and from the telephone terminal based on 10BaseT or 10/100BaseT Ethernet running TCP/IP protocols. Some IP telephones may be equipped with multiple RJ-45 Ethernet connector ports and an integrated Ethernet hub/switch to support connections to the customer premises LAN and desktop PC clients. Newer IP telephones also may be designed with a USB connector port.

Basic IP telephone software modules include a variety of user interface drivers (display, keypad, ringer, user procedures), voice processing modules, telephony signaling gateway modules, network management modules, and system service modules. The voice processing software modules include a PCM interface unit; a tone generator (call progress tones, in-band DTMF signaling digits); a line echo canceler unit (ITU G.168-compliant echo cancellation on sampled, full-duplex voice port signals); an acoustic echo canceler for terminals equipped with a speakerphone; VAD; voice codec unit (compression and packeting of the 64-Kbps digital stream received from the station user based on a variety of algorithms, such as G.711, G.723.1, G.729/A, etc.); packet playout unit (compensation for network delay, jitter, and packet loss); packet protocol encapsulation unit (based on RTP, which runs directly on top of the UDP); voice encryption (to ensure privacy); and a control unit (coordinates the exchange of monitor and control information between the voice processing module and the telephony signaling and network management modules).

The telephony signaling gateway subsystem in an IP telephone performs the basic functions for call setup and teardown procedures. Software modules used by this subsystem include call processing, address translation and parsing, and network signaling. The most widely implemented network signaling standard used by currently available IP-PBX systems is H.323 protocol. H.323 is an ITU standard that defines several signaling and protocol specifications for multimedia communications between LAN-based terminals and network equipment. The main H.323 standards used for VoIP in an IP telephone are H.225–Call Signaling Protocol (based on Q.931), H.245–Control Protocol; RAS Protocol; and RTCP. An emerging network signaling standard not currently used by any commercially available IP-PBX, but being planned for by most suppliers, is SIP. SIP is the protocol developed and promoted by the Internet Engineering Task Force (IETF) and is forecasted to be widely implemented in network hosted services, such as IP-Centrex, and may eventually replace H.323 as the primary signaling protocol used by premises communications systems.