User's Manual

Chapter 21 Voice 188
1 User
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.
2 Pr
oxy 1 sends a SIP INVITE request to Proxy 2. Proxy 2 sends a response indicating that it is trying
to complete the request.
3 Pr
oxy 2 sends a SIP INVITE request to User Agent 2.
4 User
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.
5 User
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.
6 User
Agent 1 and User Agent 2 exchange RTP packets containing voice data directly, without
involving the proxies.
7 Wh
en User Agent 2 hangs up, he sends a BYE request.
8 User
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 Router supports the following codecs.
• G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal
amplitu
des 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) wa
veform 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 Analysis-by-Synthesis (AbS) hybrid
waveform codec that uses a filter based on
information about how the human vocal tract produces sounds. G.729 provides good sound
quality and reduces the required bandwidth to 8 kbps.
BYE
200 OK
Table 105 SIP Call Progression
UA 1 PROXY 1 PROXY 2 UA 2