| Codecs-Digitized Voice standards |
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Speech is carried in packets that use the Real Time Protocol (RTP) standard. Each RTP packet contains a piece of a digitized word. Multiple RTP packets, when combined at the receiving IP phone, produce a spoken word. The RTP IP PBX vendors commonly implement RTP. Proprietary protocols that operate like RTP are uncommon. The speech paths connect directly between phones and gateways.
There are several voice digitization standards and some proprietary techniques in use. Most vendors support one or more of the following ITU standards and avoid proprietary solutions: • G.711 is the default standard for all vendors, as well as for the PSTN. This standard digitizes voice into 64 Kbps and does not compress the voice • G.729 is supported by many vendors for compressed voice operating at 8 Kps. It is one of the most commonly implemented standards. • G.723.1 was once the recommended compression standard. It operates at 6.3 and 5.3 Kbps. Although this standard reduces bandwidth consumption, voice is noticeably poorer than with G.729 and is not very popular for VoIP. • G.722 operates at 64 Kbps but offers high-fidelity speech. Whereas, the three previously described standards deliver an analog sound range of 3.4 kHz, G.722 delivers 7kHz. This version of digitized speech will become common in the future In all cases, the IP phones and gateways collect about 10 to 30ms of digital speech and place it in the RTP packet for transmission. |