3Com MSR 50 Series, Руководство по настройке

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SIP O
VERVIEW
When configuring SIP, go to these sections for information you are interested in:
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“Introduction to SIP” on page 2367
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“SIP UA Configuration” on page 2373
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“Displaying and Maintaining SIP UAs” on page 2379
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“SIP UA Configuration Examples” on page 2379
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“Troubleshooting” on page 2382
Introduction to SIP
Session initiation protocol (SIP) is an application layer control protocol that
establishes, modifies, and terminates multimedia sessions such as IP phone calls,
multimedia distribution, and multimedia conferences. It is the core component in
the multimedia data and control architecture of IETF and the RFC for it is RFC3261.
SIP involves issues such as signaling control in IP networks and communication
with soft switch platforms, intending to build a next generation value-added
service platform to deliver better value-added services to telecom carriers, banks,
and financial organizations.
SIP is used for initiating sessions. It sets up and terminates a multimedia session
involving a group of participants and dynamically adjusts and modifies session
characteristics such as required session bandwidth, media type (voice, video, or
data), media encoding/decoding format, and multicast/unicast. SIP is based on text
encoding and constructed taking HTTP, a quite mature protocol, as a model. Easy
to extend and implement, it is suitable for implementing Internet-based
multimedia conference systems.
SIP adopts the Client/Server model and sets up user calls through communication
between user agents and proxy servers.
As an application layer protocol, SIP uses UDP for transport normally.
This section covers these topics:
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“Terms” on page 2368
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“Functionality and Features of SIP” on page 2369
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“SIP Messages” on page 2370
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“SIP Fundamentals” on page 2370