vity (3) voice packets detection compression canceler Figure 15.5 A simplified functional block diagram of a gateway providing and interface between a PCM bit stream deriving from the PSTN on the left and an IP network. the content. A gateway senses the presence of data by the presence of a 2100-Hz tone in the time slot. The next signal type in the time slot looks for its DTMF signaling tones (see Chapter 4). If there is no modem tone nor DTMF tones in the time slot, then the gateway assumes the time slot contains human speech. Three actions now have to be accomplished. "Silence" is removed; the standard PCM compression algorithm is applied; an echo canceler is switched in. There are three digital formats used for voice-over packet: 1. Internet Protocol (IP) (Chapter 12, Section 12.3). 2. Frame Relay (Chapter 12, Section 12.5). 3. ATM-Asynchronous Transfer Mode (Chapter 20). 15.4.2 An IP Packet as Used for VolP Assume for argument that we use either a G.711 or G.726 packet (Table 15.1) IP packet. The packet consists of a header and a payload. Figure 15.6 shows a typical IP packet. Of interest, as one may imagine, is its payload. In the case of G.711 (standard PSTN PCM), there may be a transmission rate of 100 packets per second with 80 bytes in the payload of each packet. Of course our arithmetic comes out just right and we get 8000 samples per second, the Nyquist sampling rate for a 4-kHz analog voice channel. Another transmission rate for G.711 is 50 packets per second where each packet will have 160 bytes, again achieving 8000 samples per second per voice channel. The total raw bytes (octets) per channel come out as follows: 40 bytes for layers 3 and 4 overhead (IP), plus 8 bytes for layer 2 (link layer) overhead. So we add 48 to 80 or 160 bytes (from the previous paragraph) and we get 128 or 208 bytes for a raw packet. The efficiency is nothing to write home about. Keep in mind that the primary concern of the VoIP designer is delay. 15.4.3 The Delay Tradeoff Human beings are intolerant of delay on a full-dupl