How Is VoIP Converted to TDM?
A Time-Division Multiplexing (TDM) voice channel is a 64-kbps data stream that encodes the voice according to the rules of Pulse Code Modulation (PCM). Voice over Internet Protocol (VoIP) can be encoded in several ways. The coder/decoded, called a codec, determines how the voice is digitized.
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Codecs
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The codec used determines the pattern of bits used to represent the voice. The codec that digitizes voice using PCM is the G.711 codec. VoIP codecs include G.729 and G.723.1. Each codec analyzes a small sample of the voice and encodes it. The encoded represents the sound as received by the phone. Each codec uses different rules to encode the voice.
Real Time Protocol
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The Real Time Protocol contains timing information to allow the bits to be converted back to real-world signals. The digitized sample produced by the codec is put inside a Real Time Protocol (RTP) header. The RTP header includes the timing information required by the remote end to play the sound out with the correct timing. The RTP header also includes a Payload Type field to tell the remote end which codec was used to create the voice data.
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Converting to TDM
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The TDM signal can be converted back to real sound. At the destination, the Payload Type field in the RTP header determines the codec type. The timing information is also extracted from the RTP header. The output of the codec at the receiving is a 64-kbps, PCM data stream, which can be transmitted over a TDM network.
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References
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