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Chapter 11 Voice
P-2602HWLNI User’s Guide
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11.2.3 RTP
When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is used to
handle voice data transfer. See RFC 1889 for details on RTP.
11.2.4 Pulse Code Modulation
Pulse Code Modulation (PCM) measures analog signal amplitudes at regular time intervals
and converts them into bits.
11.2.5 SIP Call Progression
The following figure displays the basic steps in the setup and tear down of a SIP call. A calls
B.
A sends a SIP INVITE request to B. This message is an invitation for B to participate in a SIP
telephone call.
4 B sends a response indicating that the telephone is ringing.
5 B sends an OK response after the call is answered.
6 A then sends an ACK message to acknowledge that B has answered the call.
7 Now A and B exchange voice media (talk).
8 After talking, A hangs up and sends a BYE request.
9 B replies with an OK response confirming receipt of the BYE request and the call is
terminated.
11.2.6 SIP Call Progression Through Proxies
Usually, the SIP UAC sets up a phonecall by sending a request to the SIP proxy server. Then,
the proxy server looks up the destination to which the call should be forwarded (according to
the URI requested by the SIP UAC). The request may be forwarded to more than one proxy
server before arriving at its destination.
The response to the request goes to all the proxy servers through which the request passed, in
reverse sequence. Once the session is set up, session traffic is sent between the UAs directly,
bypassing all the proxy servers in between.
Table 58 SIP Call Progression
A B
1. INVITE
2. Ringing
3. OK
4. ACK
5.Dialogue (voice traffic)
6. BYE
7. OK