Chapter 9 Voice
1User Agent 1 sends a SIP INVITE request to Proxy 1. This message is an invitation to User Agent 2 to participate in a SIP telephone call. Proxy 1 sends a response indicating that it is trying to complete the request.
2Proxy 1 sends a SIP INVITE request to Proxy 2. Proxy 2 sends a response indicating that it is trying to complete the request.
3Proxy 2 sends a SIP INVITE request to User Agent 2.
4User Agent 2 sends a response back to Proxy 2 indicating that the phone is ringing. The response is relayed back to User Agent 1 via Proxy 1.
5User Agent 2 sends an OK response to Proxy 2 after the call is answered. This is also relayed back to User Agent 1 via Proxy 1.
6User Agent 1 and User Agent 2 exchange RTP packets containing voice data directly, without involving the proxies.
7When User Agent 2 hangs up, he sends a BYE request.
8User Agent 1 replies with an OK response confirming receipt of the BYE request, and the call is terminated.
Voice Coding
A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital signals back into analog voice signals. The
•G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal amplitudes at regular time intervals and converts them into digital samples. G.711 provides very good sound quality but requires 64 kbps of bandwidth.
•G.726 is an Adaptive Differential PCM (ADPCM) waveform codec that uses a lower bitrate than standard PCM conversion. ADPCM converts analog audio into digital signals based on the difference between each audio sample and a prediction based on previous samples. The more similar the audio sample is to the prediction, the less space needed to describe it. G.726 operates at 16, 24, 32 or 40 kbps.
•G.729 is an
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