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This chapter is from the book

Voice Protocols and Codecs

Rounding out this introduction of voice and unified communications concepts, we now take a look at which protocols are used that provide the audio, video, and data experience. So far, I have mentioned SIP, which is used as the protocol for audio, video, and data. Another key protocol used by VoIP that is important to understand is Real-time Transport Protocol (RTP). RTP works with both H.323 and SIP, explained earlier in this chapter, and was originally created as the standard protocol used for VoIP. RTP is the delivery mechanism that carries audio and video over an IP network as well as the Internet.

In addition to VoIP protocols, as mentioned earlier in the QoS/QoE discussion, codecs are used to provide the actual band of voice communication. The most popular of which, and Microsoft designed, Real-time Audio (RTAudio) is a wide-band speech codec used by Microsoft’s voice communications products to compress the speech/audio used in a multiperson or two-way conversation (see Figure 1.11).

Figure 1.11

Figure 1.11 An example of RTAudio sample rates

This level of audio compression gives Microsoft the competitive advantage over traditional Telco providers in that the audio compression provided by the RTAudio codec ensures nonpacket loss, which ultimately means less or no jitter on the line. RTAudio is susceptible to delays, though, which is why QoS and QoE are used to overcompensate for any flaws that may be present within this voice communication infrastructure.

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