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Symbian3/SDK/Source/GUID-093516F2-9076-5A96-9BEF-F5A77F75239C.dita
changeset 0 89d6a7a84779 
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/Symbian3/SDK/Source/GUID-093516F2-9076-5A96-9BEF-F5A77F75239C.dita	Thu Jan 21 18:18:20 2010 +0000
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+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (c) 2007-2010 Nokia Corporation and/or its subsidiary(-ies) All rights reserved. -->
+<!-- This component and the accompanying materials are made available under the terms of the License 
+"Eclipse Public License v1.0" which accompanies this distribution, 
+and is available at the URL "http://www.eclipse.org/legal/epl-v10.html". -->
+<!-- Initial Contributors:
+    Nokia Corporation - initial contribution.
+Contributors: 
+-->
+<!DOCTYPE concept
+  PUBLIC "-//OASIS//DTD DITA Concept//EN" "concept.dtd">
+<concept id="GUID-093516F2-9076-5A96-9BEF-F5A77F75239C" xml:lang="en"><title>SIP
+Overview</title><prolog><metadata><keywords/></metadata></prolog><conbody>
+<section id="GUID-EFBD7A9F-1F43-480E-B122-BFA457674AAC"><title>Purpose</title> <p>The Multimedia Protocols Session Initiation
+Protocol (SIP) framework supports SIP, Session Description Protocol (SDP),
+and Signaling Compression (SigComp). SIP is an application layer signaling
+protocol that creates, modifies, and terminates sessions with the users. This
+overview describes the use of SIP as defined in <xref href="http://www.ietf.org/rfc/rfc3261.txt?number=3261" scope="external">RFC 3261</xref>. It provides the information about the use
+of SDP, SigComp and also the implementation-specific mechanisms like the Client
+Resolver and Profile Agent engines. </p> </section>
+<section id="GUID-06DB8B97-9395-4BA6-AA42-6E630B3D1244"><title>Required background</title> <p>This overview does not describe
+how the SIP protocol works, but assumes that the user knows about SIP, SDP,
+and associated protocols that set up multimedia sessions for use cases such
+as VoIP. </p> </section>
+<section id="GUID-1C54AC1A-F2F3-40B9-A4B3-855EAA3D0870"><title>Key concepts</title> <dl>
+<dlentry>
+<dt>Session Initiation Protocol</dt>
+<dd><p>SIP is an application layer signaling protocol used to create, modify,
+and terminate sessions with the users. The SIP subsystem architecture is based
+on the generic Session Initiation Protocol, <xref href="http://www.ietf.org/rfc/rfc3261.txt?number=3261" scope="external">RFC 3261</xref>. It indicates that the 3GPP IMS specific
+SIP message flows. SIP message semantics are the responsibility of the clients,
+with the 3GPP Release 5 IMS specific security mechanism implemented as an
+ECOM plug-in. </p> </dd>
+</dlentry>
+</dl> <p>The following components or subsystems are part of the Symbian SIP
+framework: </p> <ul>
+<li id="GUID-122A91B3-55E3-5521-B1B3-065F5E744A21"><dl>
+<dlentry>
+<dt>Session Description Protocol (SDP)</dt>
+<dd><p>SDP provides a standard representation for multimedia information and
+a standard method to transport that information. The SDP is implemented as
+an independent component and is used with transport mechanisms such as SIP.
+The SDP Codec subsystem provides services to convert SDP descriptions to internal
+representation and internal representations to SDP descriptions. SDP provides
+an API for its clients to encode or decode SDP descriptions and to set or
+get the SDP field values. For more information about the APIs supported by
+the SDP Codec subsystem, see <xref href="GUID-28636BB0-2C25-5D6C-9777-41AA4F3D8F31.dita">SDP
+Codec API</xref>. </p> </dd>
+</dlentry>
+</dl> </li>
+<li id="GUID-809DA7AA-1AFB-52AC-AA67-EDD218C34831"><dl>
+<dlentry>
+<dt>Signaling Compression</dt>
+<dd><p>The Signaling Compression component provides services to compress and
+decompress the text based protocols such as SIP, SDP, and Real Time Session
+Protocol (RTSP). Compressors that use different compression algorithms are
+dynamically determined and loaded at run-time using the ECOM architecture.
+UDVM provides the decompression functionality and runs the uploaded decompression
+algorithms. The component architecture allows new compressors that use a plug-in
+mechanism and static dictionaries to be included in the implementation. </p> </dd>
+</dlentry>
+</dl> </li>
+<li id="GUID-C2040FE1-CC4A-52AB-8832-C6D6B64D52F0"><dl>
+<dlentry>
+<dt>SIP Client Resolver</dt>
+<dd><p>The SIP Client Resolver defines the architecture for resolving or identifying
+the target client when it receives the SIP request from a default port. For
+information about the SIP Client Resolver API, see <xref href="GUID-A3D1684D-1DF1-5CAF-A0BB-A32685528596.dita">SIP
+Client Resolver API</xref>. </p> </dd>
+</dlentry>
+</dl> </li>
+</ul> </section>
+<section id="GUID-981591E6-A9E0-4576-8CD3-FA5F1D445FDD"><title>API summary</title> <p>The following are some of the main
+APIs used in SIP communication. </p> <table id="GUID-F4063A91-4F8E-5965-9DBA-9CA72AC4A6BC">
+<tgroup cols="2"><colspec colname="col0"/><colspec colname="col1"/>
+<thead>
+<row>
+<entry>API name</entry>
+<entry>Description</entry>
+</row>
+</thead>
+<tbody>
+<row>
+<entry><p> <xref href="GUID-C5CBB74D-0EC0-3E22-8CF2-B3F61E1EB79E.dita"><apiname>SIP Client API</apiname></xref>  </p> </entry>
+<entry><p>An API that manages SIP transactions and SIP dialogs. It sets the
+security mechanism parameters, receives connection state change events, queries
+supported security mechanism and signaling compression support. For more information,
+see <xref href="GUID-E339A683-94E2-54E8-BB2B-286B11B0FCE3.dita">SIP Client API</xref>. </p> </entry>
+</row>
+<row>
+<entry><p> <xref href="GUID-3433DBD2-6444-32C7-88F9-EFE0629EC3F3.dita"><apiname>SIP Codec API</apiname></xref>  </p> </entry>
+<entry><p>An API that provides services for managing SIP requests and responses.
+This means it sets and obtains the headers and field values. For more information,
+see <xref href="GUID-CD0EF6F2-9515-5884-B979-C3C391B7DA07.dita">SIP Codec API</xref>. </p> </entry>
+</row>
+<row>
+<entry><p> <xref href="GUID-390AECC6-C8FF-34ED-8717-7BFCC4CF92BE.dita"><apiname>SDP Codec API</apiname></xref>  </p> </entry>
+<entry><p>An API that provides a mechanism to encode or decode SDP messages.
+For more information, see <xref href="GUID-28636BB0-2C25-5D6C-9777-41AA4F3D8F31.dita">SDP
+Codec API</xref>. </p> </entry>
+</row>
+<row>
+<entry><p> <xref href="GUID-A346DFC5-5528-342D-89E2-A9A710AB377F.dita"><apiname>SIP Profile API</apiname></xref>  </p> </entry>
+<entry><p>An API that is used to get SIP profile parameters. If a profile
+is not configured for automatic registration it registers or deregisters the
+specific SIP profile. It also creates, uses, or terminates SIP transactions
+and dialogs using profile information. For more information, see <xref href="GUID-399EAC59-643D-5DED-AB67-9F021CA5687A.dita">SIP
+Profile API</xref>. </p> </entry>
+</row>
+<row>
+<entry><p> <xref href="GUID-8696D110-FF44-3ABA-B9F8-C314B8AFB6AC.dita"><apiname>SIP Client Resolver API</apiname></xref>  </p> </entry>
+<entry><p>An ECOM plug-in API that is used by applications that use the SIP
+stack to receive out of dialog incoming request, for example an Incoming VoP
+call. For more information, see <xref href="GUID-A3D1684D-1DF1-5CAF-A0BB-A32685528596.dita">SIP
+Client Resolver API</xref>. </p> </entry>
+</row>
+<row>
+<entry><p> <xref href="GUID-09F69A21-5F66-3344-BD04-69D02555E9F1.dita"><apiname>SIP High Level API</apiname></xref>  </p> </entry>
+<entry><p>An API that encapsulates the SIP call flows inside the <xref href="GUID-BED8A733-2ED7-31AD-A911-C1F4707C67FD.dita"><apiname>RConnection</apiname></xref> and <xref href="GUID-0AFDA357-EE44-3788-9CAB-162B874134BF.dita"><apiname>RSubConnection</apiname></xref> APIs.
+For more information, see <xref href="GUID-4120651F-E0B9-5927-96B9-2662C51F5A09.dita">SIP
+High Level API</xref>. </p> </entry>
+</row>
+</tbody>
+</tgroup>
+</table> </section>
+<section id="GUID-8EB2C029-3C84-406A-BB2B-FD76398B89D8"><title>Typical uses</title> <p><b>When to use which API</b> </p><p>The
+following table describes when to use which API. </p><table id="GUID-AA11D59D-4373-4878-82A5-BFB2C8C3C336">
+<tgroup cols="2"><colspec colname="col1"/><colspec colname="col2"/>
+<thead>
+<row>
+<entry valign="top"><p>API name</p></entry>
+<entry valign="top"><p>Usage</p></entry>
+</row>
+</thead>
+<tbody>
+<row>
+<entry><p>SIP Client API </p></entry>
+<entry><p>Use the SIP Client API to send and receive SIP messages, create
+registrations, form and tear down dialogs initiated by INVITE, REFER and SUBSCRIBE
+requests. For more information, see SIP Client API. </p></entry>
+</row>
+<row>
+<entry><p>SIP Codec API </p></entry>
+<entry><p>Use the SIP Codec API to encode and decode SIP headers and their
+fields. For more information, see SIP Codec API.</p></entry>
+</row>
+<row>
+<entry><p>SDP Codec API </p></entry>
+<entry><p>Use the SDP Codec API to encode SDP descriptions into an internal
+form, decode the descriptions from an internal form, set and get the SDP field
+values. For more information, see SDP Codec API.</p></entry>
+</row>
+<row>
+<entry><p>SIP Profile API </p></entry>
+<entry><p>Use the SIP Profile API to register through a profile. For more
+information, see SIP Profile API.</p></entry>
+</row>
+<row>
+<entry><p>SIP Client Resolver API </p></entry>
+<entry><p>Use the SIP Client Resolver API to resolve or identify
+the target client when SIP requests are received from the default port. For
+more information, see SIP Client Resolver API. </p></entry>
+</row>
+<row>
+<entry><p>SIP High Level API </p></entry>
+<entry><p>Use the SIP High Level API to register with the SIP server, invite
+a session and subscribe to events. For more information, see SIP High Level
+API.</p></entry>
+</row>
+</tbody>
+</tgroup>
+</table> </section>
+<section id="GUID-A373C1D8-0678-4A45-BAAC-CA37324A4426"><title>See also</title> <ul>
+<li id="GUID-0E00CD92-FCD1-5EFE-87E0-847D3216401F"><p> <xref href="http://www.ietf.org/rfc/rfc3261.txt?number=3261" scope="external">SIP: Session Initiation Protocol, RFC 3261</xref>  </p> </li>
+<li id="GUID-F8278257-C4CE-5FBD-9DCB-C7B47A8C5273"><p> <xref href="http://www.3gpp.org/" scope="external">3GPP TS 24.228 v.5.2.0 Signaling flows for the IP multimedia
+call control based on SIP and SDP; Stage 3</xref>  </p> </li>
+<li id="GUID-F7AF8C36-7D6D-557D-91BA-015DE811FDEA"><p> <xref href="http://www.3gpp.org/" scope="external"> 3GPP TS 24.229 v.5.4.0 Internet Protocol (IP) multimedia
+call control protocol based on Session Initiation Protocol (SIP) and Session
+Description Protocol (SDP); Stage 3</xref>  </p> </li>
+</ul> </section>
+</conbody></concept>
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FCL/sftools/depl/docscontent