telephoneevents.sgml

linphone源码-1.3.5.tar.gz所需要的库文件

<refentry id="ortp-Telephone-events---RFC2833-">
<refmeta>
<refentrytitle>Telephone events (RFC2833)</refentrytitle>
<manvolnum>3</manvolnum>
<refmiscinfo>ORTP Library</refmiscinfo>
</refmeta>

<refnamediv>
<refname>Telephone events (RFC2833)</refname><refpurpose></refpurpose>
</refnamediv>

<refsynopsisdiv><title>Synopsis</title>

<synopsis>

#include &lt;telephonyevents.h&gt;


struct      <link linkend="telephone-event">telephone_event</link>;
<link linkend="gint">gint</link>        <link linkend="rtp-session-telephone-events-supported">rtp_session_telephone_events_supported</link>
                                            (<link linkend="RtpSession">RtpSession</link> *session);
<link linkend="mblk-t">mblk_t</link>*     <link linkend="rtp-session-create-telephone-event-packet">rtp_session_create_telephone_event_packet</link>
                                            (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="int">int</link> start);
<link linkend="gint">gint</link>        <link linkend="rtp-session-add-telephone-event">rtp_session_add_telephone_event</link> (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="mblk-t">mblk_t</link> *packet,
                                             <link linkend="guchar">guchar</link> event,
                                             <link linkend="gint">gint</link> end,
                                             <link linkend="guchar">guchar</link> volume,
                                             <link linkend="guint16">guint16</link> duration);
<link linkend="gint">gint</link>        <link linkend="rtp-session-read-telephone-event">rtp_session_read_telephone_event</link>
                                            (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="mblk-t">mblk_t</link> *packet,
                                             <link linkend="telephone-event-t">telephone_event_t</link> **tab);
<link linkend="gint">gint</link>        <link linkend="rtp-session-send-dtmf">rtp_session_send_dtmf</link>           (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="gchar">gchar</link> dtmf,
                                             <link linkend="guint32">guint32</link> userts);
</synopsis>
</refsynopsisdiv>









<refsect1>
<title>Description</title>
<para>

</para>
</refsect1>

<refsect1>
<title>Details</title>
<refsect2>
<title><anchor id="telephone-event">struct telephone_event</title>
<indexterm><primary>telephone_event</primary></indexterm><programlisting>struct telephone_event {
#ifdef WORDS_BIGENDIAN
	guint32 event:8;
	guint32 E:1;
	guint32 R:1;
	guint32 volume:6;
	guint32 duration:16;
#else
	guint32 event:8;
	guint32 volume:6;
	guint32 R:1;
	guint32 E:1;
	guint32 duration:16;
#endif
};
</programlisting>
<para>

</para></refsect2>
<refsect2>
<title><anchor id="rtp-session-telephone-events-supported">rtp_session_telephone_events_supported ()</title>
<indexterm><primary>rtp_session_telephone_events_supported</primary></indexterm><programlisting><link linkend="gint">gint</link>        rtp_session_telephone_events_supported
                                            (<link linkend="RtpSession">RtpSession</link> *session);</programlisting>
<para>
Tells whether telephony events payload type is supported within the context of the rtp
session.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara>	a rtp session 
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> the payload type number used for telephony events if found, -1 if not found.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-create-telephone-event-packet">rtp_session_create_telephone_event_packet ()</title>
<indexterm><primary>rtp_session_create_telephone_event_packet</primary></indexterm><programlisting><link linkend="mblk-t">mblk_t</link>*     rtp_session_create_telephone_event_packet
                                            (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="int">int</link> start);</programlisting>
<para>
Allocates a new rtp packet to be used to add named telephony events. The application can use
then <link linkend="rtp-session-add-telephone-event"><function>rtp_session_add_telephone_event()</function></link> to add named events to the packet.
Finally the packet has to be sent with <link linkend="rtp-session-sendm-with-ts"><function>rtp_session_sendm_with_ts()</function></link>.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>start</parameter>&nbsp;:</term>
<listitem><simpara>	boolean to indicate if the marker bit should be set.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> a message block containing the rtp packet if successfull, NULL if the rtp session
cannot support telephony event (because the rtp profile it is bound to does not include
a telephony event payload type).
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-add-telephone-event">rtp_session_add_telephone_event ()</title>
<indexterm><primary>rtp_session_add_telephone_event</primary></indexterm><programlisting><link linkend="gint">gint</link>        rtp_session_add_telephone_event (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="mblk-t">mblk_t</link> *packet,
                                             <link linkend="guchar">guchar</link> event,
                                             <link linkend="gint">gint</link> end,
                                             <link linkend="guchar">guchar</link> volume,
                                             <link linkend="guint16">guint16</link> duration);</programlisting>
<para>
Adds a named telephony event to a rtp packet previously allocated using
<link linkend="rtp-session-create-telephone-event-packet"><function>rtp_session_create_telephone_event_packet()</function></link>.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara>	a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>packet</parameter>&nbsp;:</term>
<listitem><simpara>	a rtp packet as a <link linkend="mblk-t"><type>mblk_t</type></link>
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>event</parameter>&nbsp;:</term>
<listitem><simpara>	the event type as described in rfc2833, ie one of the TEV_ macros.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>end</parameter>&nbsp;:</term>
<listitem><simpara>		boolean to indicate if the end bit should be set. (end of tone)
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>volume</parameter>&nbsp;:</term>
<listitem><simpara>	the volume of the telephony tone, as described in rfc2833
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>duration</parameter>&nbsp;:</term>
<listitem><simpara>the duration of the telephony tone, in timestamp unit.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara>0 on success.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-read-telephone-event">rtp_session_read_telephone_event ()</title>
<indexterm><primary>rtp_session_read_telephone_event</primary></indexterm><programlisting><link linkend="gint">gint</link>        rtp_session_read_telephone_event
                                            (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="mblk-t">mblk_t</link> *packet,
                                             <link linkend="telephone-event-t">telephone_event_t</link> **tab);</programlisting>
<para>
Reads telephony events from a rtp packet. *<parameter>tab</parameter> points to the beginning of the event buffer.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session from which telephony events are received.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>packet</parameter>&nbsp;:</term>
<listitem><simpara>	a rtp packet as a mblk_t.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>tab</parameter>&nbsp;:</term>
<listitem><simpara>		the address of a pointer.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> the number of events in the packet if successfull, 0 if the packet did not
contain telephony events.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-send-dtmf">rtp_session_send_dtmf ()</title>
<indexterm><primary>rtp_session_send_dtmf</primary></indexterm><programlisting><link linkend="gint">gint</link>        rtp_session_send_dtmf           (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="gchar">gchar</link> dtmf,
                                             <link linkend="guint32">guint32</link> userts);</programlisting>
<para>
This functions creates telephony events packets for <parameter>dtmf</parameter> and sends them.
It uses <link linkend="rtp-session-create-telephone-event-packet"><function>rtp_session_create_telephone_event_packet()</function></link> and
<link linkend="rtp-session-add-telephone-event"><function>rtp_session_add_telephone_event()</function></link> to create them and finally
<link linkend="rtp-session-sendm-with-ts"><function>rtp_session_sendm_with_ts()</function></link> to send them.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>dtmf</parameter>&nbsp;:</term>
<listitem><simpara> a character meaning the dtmf (ex: '1', '#' , '9' ...)
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>userts</parameter>&nbsp;:</term>
<listitem><simpara> the timestamp
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara>	0 if successfull, -1 if the session cannot support telephony events or if the dtmf
given as argument is not valid.
</simpara></listitem></varlistentry>
</variablelist></refsect2>

</refsect1>




</refentry>