--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/Symbian3/SDK/Source/GUID-0ACAFB87-0D45-5BAA-B40A-9A7DC1646B01.dita	Wed Mar 31 11:11:55 2010 +0100
@@ -0,0 +1,85 @@
+<?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-0ACAFB87-0D45-5BAA-B40A-9A7DC1646B01" xml:lang="en"><title>Managing
+the RTCP packets and reports</title><prolog><metadata><keywords/></metadata></prolog><conbody>
+<p>This section describes how to send various types of RTCP packets, set the
+stream parameters, get stream statistics and close the session. </p>
+<section id="GUID-663C6781-3137-5D48-A836-6ADDD4474805"><title>Sending various
+RTCP packets</title><p>To automatically send SR, RR, or SDES packets, call <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-B32E14CD-DE13-3E89-96B4-5713CB7217A7"><apiname>RRtpSession::SetRTCPAutoSend()</apiname></xref> with
+its paramter set to ETrue. For more information about these packets, see Managing
+send and receive reports.</p><codeblock xml:space="preserve">session.SetRTCPAutoSend(ETrue);
+</codeblock><p>Set the RTCP auto-send time interval in microseconds by calling <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-506A9D47-03DE-36D0-8466-D7BFB725FBB1"><apiname>RRtpSession::SetRtcpInterval()</apiname></xref>.</p><codeblock xml:space="preserve">session.SetRtcpInterval(1000);
+</codeblock><p>If auto-sending is set to EFalse, call <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-7AF454C8-6943-3986-BC82-CF6BFB27AF7A"><apiname>RRtpSession::SendRTCPPacketL()</apiname></xref> to
+send a packet.</p><codeblock xml:space="preserve">//aPacket is a valid RTCP packet session.SendRTCPPacketL(aPacket);
+</codeblock><p>For more information, see the definition of TRtpSendPacketType.
+ </p><codeblock xml:space="preserve">Call RRtpSession::SendAppL() to send an APP packet.</codeblock><p>For
+an SDES RTP packet, call <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-1D4254AC-043A-3BA7-ABF7-B008DC9F4D01"><apiname>RRtpSession::SetSDESL()</apiname></xref> to set
+the SDES details such as CNAME, NAME, PHONE, EMAIL and so on. You must set
+these details before the packet is sent. The CNAME must be unique for an RTP
+session.   <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-0B04130D-585C-31D9-A1F0-B4166BC26A92"><apiname>RRtpSession::GetSDES()</apiname></xref> enables you to fetch
+the data associated with the specified SDES item.  </p><p> You can set the
+data associated with a PRIV (Private Extensions) SDES item by calling <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-D7E02919-E39A-3CA0-A8C7-BE77504C0948"><apiname>RRtpSession::SetPRIVL()</apiname></xref>.
+ PRIV (Private Extensions) SDES is not supported.   Call <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-00D4DBEA-3EDB-3F88-90B8-374D2CECB383"><apiname>RRtpSession::ByeL()</apiname></xref> to
+send an RTCP Bye packet.  </p> </section>
+<section><title>Setting the stream parameters </title> <p>The <xref href="GUID-8420B7F8-7BE3-30C0-BFC4-05F7A13361B6.dita"><apiname>SetRtpStreamParameters()</apiname></xref> method
+sets the following stream parameters: </p> <ul>
+<li id="GUID-60CD5E08-6829-5675-AC64-6CAA3C6D53D9"><p>maximum number of sequential
+packets that must be received for a good stream. </p> </li>
+<li id="GUID-8ED20A24-B6D0-5CE4-A115-459FED0E885C"><p>maximum number of dropped
+packets to be considered a dropout. </p> </li>
+<li id="GUID-688E9BB0-96E9-5090-B80F-1E75D27D2FCD"><p>maximum number of packets
+by which a packet can be delayed to be considered as dropped. </p> </li>
+</ul> <codeblock id="GUID-601FA412-12EA-571F-913C-7F4E383EE8BD" xml:space="preserve">SetRtpStreamParameters(1,100,3000);// set MinSequential, MaxMisorder &amp; MaxDropout </codeblock> </section>
+<section><title>Getting the stream statistics</title> <p>The <xref href="GUID-97FC8566-E4AD-3881-8669-134798A88779.dita#GUID-97FC8566-E4AD-3881-8669-134798A88779/GUID-63B8D608-D078-3F8E-9ABD-2CD29F1396F5"><apiname>RRtpSession::RtcpStatistics()</apiname></xref> method
+provides sending and receiving statistics about the current active RTP session.
+The statistics depend on the contents of the latest Sender Report (SR) packets
+and Receiver Report (RR) packets. The returned <xref href="GUID-53671562-A0BB-379D-AC1A-0006F70A4513.dita"><apiname>TRtcpStatistics</apiname></xref> structure
+contains the following statistics: </p> <table id="GUID-82B41FFA-00C1-5477-9727-2C3DBCA754D5">
+<tgroup cols="2"><colspec colname="col0"/><colspec colname="col1"/>
+<thead>
+<row>
+<entry>Direction </entry>
+<entry>Statistic </entry>
+</row>
+</thead>
+<tbody>
+<row>
+<entry><p>Sending </p> </entry>
+<entry><p>Total number of packets sent from the beginning of the RTP session
+to the latest SR. </p> </entry>
+</row>
+<row>
+<entry><p>Total number of payload octets sent from the beginning of the RTP
+session to the latest SR. </p> </entry>
+</row>
+<row>
+<entry><p>Receiving </p> </entry>
+<entry><p>Inter-arrival jitter. </p> </entry>
+</row>
+<row>
+<entry><p>Total number of packets lost since the beginning of the RTP session. </p> </entry>
+</row>
+<row>
+<entry><p>Fraction of packets lost since the previous SR or RR. </p> </entry>
+</row>
+</tbody>
+</tgroup>
+</table> <p>For more information about these statistics, see the RTP standard
+in <xref href="http://www.ietf.org/rfc/rfc3550.txt" scope="external">RFC 3550</xref>. </p> </section>
+<section><title>Closing a session</title> <p> <codeph>Close()</codeph> closes
+the RTP session. All the send and receive streams are disconnected after this
+call. </p> <codeblock id="GUID-6723F7F1-DDBE-5912-BF72-8B0902CDE681" xml:space="preserve">session.Close(); 
+delete session;</codeblock> </section>
+<section><title>See also</title> <p><xref href="GUID-FBA52F3B-0A7A-5FE3-82CD-72F88788D432.dita#GUID-FBA52F3B-0A7A-5FE3-82CD-72F88788D432/GUID-352B33B6-D02B-5ECB-B154-6CB06C21A377">Types
+of RTCP packets</xref>  </p> <p><xref href="GUID-5287A403-1094-5682-A443-A6C2A04A1F23.dita">Creating
+and Managing an RTP session</xref>  </p> </section>
+</conbody></concept>
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