1) H.323 - ITU standart
2) MGCP - IETF st
3) SIP - IETF st
Regardless of the signaling
protocol used, a phone call has three main stages:
- call setup
- call maintenance
- call teardown.
1) Call setup
During call setup, the destination telephone number must be resolved to an IP address, where the
call request message must be sent; this is called call routing. Call admission control (CAC) is an
optional step that determines whether the network has sufficient bandwidth for the call. If bandwidth
is inadequate, CAC sends a message to the initiator indicating that the call cannot get through
because of insufficient resources. (The caller usually hears a fast busy tone.)
If call routing and CAC succeed, a call request message is sent toward the destination. If the
destination is not busy and it accepts the call, some parameters for the call must be negotiated
before voice communication begins. Following are a few of the important parameters that must be
negotiated:
■ The IP addresses to be used as the destination and source of the VoIP packets between the call
end points
■ The destination and source User Datagram Protocol (UDP) port numbers that the RTP uses at
each call end point
■ The compression algorithm (codec) to be used for the call; for example, whether G.729,
G.711, or another standard will be used
2) Call maintenance
Call maintenance collects statistics such as packets exchanged, packets lost, end-to-end delay, and
jitter during the VoIP call. The end points (devices such as IP phones) that collect this information
can locally analyze this data and display the call quality information upon request, or they can
submit the results to another device for centralized data analysis.
3)Call teardown
is simply "hanging up" and sending appropriate notification to the other end point and any control devices so that the resources can be made free.
Analog-to-digital conversion involves four major steps:
1. Sampling
2. Quantization
3. Encoding
4. Compression (optional)
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