Symbian3/SDK/Source/GUID-A81854C9-25F1-5C3B-AF3F-ECB7CFA26677.dita
author Dominic Pinkman <dominic.pinkman@nokia.com>
Tue, 20 Jul 2010 12:00:49 +0100
changeset 13 48780e181b38
parent 0 89d6a7a84779
permissions -rw-r--r--
Week 28 contribution of SDK documentation content. See release notes for details. Fixes bugs Bug 1897 and Bug 1522.

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<concept xml:lang="en" id="GUID-A81854C9-25F1-5C3B-AF3F-ECB7CFA26677"><title>Creating the receive stream</title><prolog><metadata><keywords/></metadata></prolog><conbody><p>This page describes how to create a receive source. </p> <p>The <codeph>RRtpReceiveSource</codeph> class represents the stream of data on a single SSRC (synchronised source) and delivers the RTP packets and RTCP information in the order they arrive. </p> <p> <codeph>RRtpSession</codeph> allows you to create a new receive stream. <codeph>NewReceiveSourceLC()</codeph> and <codeph>NewReceiveSourceL()</codeph> create a new receive stream and return a handle to an <codeph>RRtpReceiveSource</codeph>. </p> <section><title>Receiving the packet</title> <p> <codeph>RRtpReceivePacket</codeph> is derived from <codeph>RRtpPacket</codeph> and provides a handle to a received RTP packet. To access the packet information, use the functions listed in <xref href="GUID-46047253-4471-502D-B48E-E1990E7F7962.dita#GUID-46047253-4471-502D-B48E-E1990E7F7962/GUID-9E7AAB77-6F14-51AE-8008-B2F1C49667A5">Preparing the packet</xref>. </p> <p> <codeph>RRtpReceiveSource</codeph> provides the following functionality: </p> <p> <codeph>IsOpen</codeph> returns <codeph>ETrue</codeph> if the stream is open. </p> <p> <codeph>GetByeReason</codeph>: Retrieves the reason for terminating the session </p> <p> <codeph>GetLastApp</codeph>: Retrieves the parameters from the last APP packet. You must specify the application name, application data, and sub-type of the APP packet. </p> <p> <codeph>Packet</codeph>: Gets the packet that has just arrived, or a closed packet if there is not one. </p> <p> <codeph>GetSDES</codeph>: Gets the data associated with the SDES (Source Description) item such as CNAME, NAME and so on. </p> <p> <codeph>SSRC</codeph>: Gets the source of stream of the RTP packets. </p> <p> <codeph>Close</codeph>: Closes the receive stream and releases all its resources. </p> <p>To register for events from a stream, refer to the section <xref href="GUID-9652782C-D5A7-584F-A61C-1340D49478A2.dita">Registering for and handling events</xref>. </p> <p> <b>Note</b>: Before receiving an RTP packet you must set the bandwidth for the RTP session. The following code snippet shows how to set the bandwidth for an RTP session. </p> <codeblock id="GUID-E4210BDD-FC9C-5660-9773-FE91683994EF" xml:space="preserve">IMPORT_C void SetBandwidth(TUint aBandwidth) </codeblock> </section> <section><title>See also</title> <p> <xref href="GUID-5287A403-1094-5682-A443-A6C2A04A1F23.dita">Creating and Managing an RTP session</xref>  </p> </section> </conbody></concept>