sessionmanager.html

JMF原版英文说明文档

<A NAME="getPassiveParticipants()"><!-- --></A><H3>
getPassiveParticipants</H3>
<PRE>
public java.util.Vector <B>getPassiveParticipants</B>()</PRE>
<DL>
<DD>Returns all the passive participants. These participants will
 include the local participant and some remote participants that
 do not send any data.<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getLocalParticipant()"><!-- --></A><H3>
getLocalParticipant</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/LocalParticipant.html">LocalParticipant</A> <B>getLocalParticipant</B>()</PRE>
<DL>
<DD>Retrieves the local participant<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getAllParticipants()"><!-- --></A><H3>
getAllParticipants</H3>
<PRE>
public java.util.Vector <B>getAllParticipants</B>()</PRE>
<DL>
<DD>Returns all the participants of this session.<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getReceiveStreams()"><!-- --></A><H3>
getReceiveStreams</H3>
<PRE>
public java.util.Vector <B>getReceiveStreams</B>()</PRE>
<DL>
<DD>Returns the ReceiveStreams created by the
 SessionManager. These are streams formed when the RTPSM
 detects a new source of RTP data. 
 ReceiveStreams returned are a snapshot of the current state in the
 RTPSesionManager and the ReceiveStreamListener interface may be used
 to get notified of additional streams.<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getSendStreams()"><!-- --></A><H3>
getSendStreams</H3>
<PRE>
public java.util.Vector <B>getSendStreams</B>()</PRE>
<DL>
<DD>Returns the SendStreams created by the
 SessionManager.
 SendStreams returned are a snapshot of the current state in the
 RTPSesionManager and the SendStreamListener interface may be used
 to get notified of additional streams.<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getStream(long)"><!-- --></A><H3>
getStream</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/RTPStream.html">RTPStream</A> <B>getStream</B>(long&nbsp;filterssrc)</PRE>
<DL>
<DD>Returns the RTPStream created by the SessionManager that
 has the same SSRC as the filterssrc. If no stream with the
 filterssrc exits, null is returned<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getMulticastScope()"><!-- --></A><H3>
getMulticastScope</H3>
<PRE>
public int <B>getMulticastScope</B>()</PRE>
<DL>
<DD>Allows the user to retrieve the multicast TTL set for this
session<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="setMulticastScope(int)"><!-- --></A><H3>
setMulticastScope</H3>
<PRE>
public void <B>setMulticastScope</B>(int&nbsp;multicastScope)</PRE>
<DL>
<DD>Sets the IP Multicast TTL for this session. <P><DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Parameters:</B><DD><CODE>multicastScope</CODE> - Specifies the new multicast scope for the
 session. A multicastScope less than 1 defaults to a scope of 1
 set for the session. <P></DL>
</DD>
</DL>
<HR>

<A NAME="closeSession(java.lang.String)"><!-- --></A><H3>
closeSession</H3>
<PRE>
public void <B>closeSession</B>(java.lang.String&nbsp;reason)</PRE>
<DL>
<DD>Informs RTP that the application is ready to terminate the
 session. All open streams will be closed after this call
 completes, and the  object implementing this  interface is no longer
 useful as an SessionManager. <P><DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Parameters:</B><DD><CODE>reason</CODE> - A string that RTCP will send out to other
 participants  as the reason the local participant  has quit the
 session.This RTCP packet will go out with  the default SSRC of the
 session. If supplied as null, a default reason will be supplied
by RTP.  <P></DL>
</DD>
</DL>
<HR>

<A NAME="generateCNAME()"><!-- --></A><H3>
generateCNAME</H3>
<PRE>
public java.lang.String <B>generateCNAME</B>()</PRE>
<DL>
<DD>This function can be used to generate a CNAME using the scheme
 described in RFC1889. This function is provided in order to facilitate CNAME
 generation prior to actual stream creation. <P><DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Returns:</B><DD>The generated CNAME. <P></DL>
</DD>
</DL>
<HR>

<A NAME="generateSSRC()"><!-- --></A><H3>
generateSSRC</H3>
<PRE>
public long <B>generateSSRC</B>()</PRE>
<DL>
<DD>This function can be used to generate a SSRC using the scheme
 described in RFC1889. This function is provided in order to facilitate SSRC
 generation prior to actual stream creation, which may be useful for call
 control functions which need to exchange SSRCs before stream transmission
 commences.  <P><DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Returns:</B><DD>The generated SSRC. <P></DL>
</DD>
</DL>
<HR>

<A NAME="getSessionAddress()"><!-- --></A><H3>
getSessionAddress</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/SessionAddress.html">SessionAddress</A> <B>getSessionAddress</B>()</PRE>
<DL>
<DD>Method to get the Destination SessionAddress of this SessionManager
 This will return the data address/port and control address/port
 that the RTP manager is receiving data/control packets on and
 sending RTCP packets to.<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getLocalSessionAddress()"><!-- --></A><H3>
getLocalSessionAddress</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/SessionAddress.html">SessionAddress</A> <B>getLocalSessionAddress</B>()</PRE>
<DL>
<DD>Method to get the local SessionAddress  of this
 SessionManager. This will return the local address and port
 bound to of the socket which is sending out RTP/RTCP packets. Since the
 RTPSM does not send out any RTP packets, the destaddress/port
 field of SessionAddress will not be filled in.
 Note: current implementations of JDK will return
 0.0.0.0/0.0.0.0 for a socket.getLocalAddress() call.<DD><DL>
</DL>
</DD>
</DL>
<HR>

<A NAME="getGlobalReceptionStats()"><!-- --></A><H3>
getGlobalReceptionStats</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/GlobalReceptionStats.html">GlobalReceptionStats</A> <B>getGlobalReceptionStats</B>()</PRE>
<DL>
<DD>This method will provide access to overall data and control
 messsage reception statistics for this Session. Statistics on
 data from individual sources is available from method
 getSourceReceptionStats() of interface ReceiveStream<DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Returns:</B><DD>The GlobalReceptionStats for this session</DL>
</DD>
</DL>
<HR>

<A NAME="getGlobalTransmissionStats()"><!-- --></A><H3>
getGlobalTransmissionStats</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/GlobalTransmissionStats.html">GlobalTransmissionStats</A> <B>getGlobalTransmissionStats</B>()</PRE>
<DL>
<DD>This method will provide access to overall data and control
 messsage transmission statistics for this Session. Statistics on
 data from individual sources is available from method
 getSourceTransmissionStats() of interface SendStream<DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Returns:</B><DD>The GlobalTransmissionStats for this session</DL>
</DD>
</DL>
<HR>

<A NAME="createSendStream(int, javax.media.protocol.DataSource, int)"><!-- --></A><H3>
createSendStream</H3>
<PRE>
public <A HREF="../../../javax/media/rtp/SendStream.html">SendStream</A> <B>createSendStream</B>(int&nbsp;ssrc,
                                   <A HREF="../../../javax/media/protocol/DataSource.html">DataSource</A>&nbsp;ds,
                                   int&nbsp;streamindex)
                            throws <A HREF="../../../javax/media/format/UnsupportedFormatException.html">UnsupportedFormatException</A>,
                                   <A HREF="../../../javax/media/rtp/SSRCInUseException.html">SSRCInUseException</A>,
                                   java.io.IOException</PRE>
<DL>
<DD>This method is used to create a sending stream within the RTP
 session. For each time the call is made, a new sending stream
 will be created. This stream will use the SDES items as entered
 in the initSession() call for all its RTCP messages. Each stream
 is sent out with a new SSRC (Synchronisation SouRCe
 identifier), but from the same participant  i.e. local
 participant. <BR><DD><DL>
</DL>
</DD>
<DD><DL>
<DT><B>Parameters:</B><DD><CODE>ssrc</CODE> - The Synchronisation source identifier to be used when
 sending out this data stream. Note: Since the ssrc  to be used by
 the local member is supplied in the initSession() call as well,
 the first SendStream created will use that ssrc and hence
 IGNORE the ssrc supplied in createSendStream. Subsequent calls to
 createSendStream() will use the ssrc supplied as this argument.
 Use createSendStream (DataSource, int, int) to start a send
 stream without supplying an SSRC. <BR><DD><CODE>ds</CODE> - This is the PushOutputDataSource or
 PullOutputDataSource which is the output datasource of the
 Processor. This datasource may contain more than one
 stream. The stream which is used in creating this RTP
 stream is specified in the next parameter of stream.<BR><DD><CODE>streamindex</CODE> - The index of the sourcestream from which
 data is sent out on this RTP stream. An index of 1 would indicate the first
 sourcestream of this datasource should be used to create the RTP
 stream. If the index is set to zero, it would indicate a RTP
 mixer operation is desired. i.e. all the streams of this
 datasource must be mixed into one single stream from one single
 SSRC. <BR><DD><CODE>packet_interval</CODE> - This is the packetization interval in
 millisecs that must constitute each RTP packet when it is
 transmitted over the network. The packetization interval
 determines the  minimum end-to-end delay; longer packets
 introduce less  header overhead but higher delay and make packet loss more
 noticeable. For non interactive applications such as lectures or
 links with  severe bandwidth constraints, a higher packetization delay may be
 appropriate. <BR>
 
 If this parameter is supplied as zero, the  
 packetizer's default packetization interval will be used.<BR>