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<a name="FsRtpSession"></a><div class="titlepage"></div>
<div class="refnamediv"><table width="100%"><tr>
<td valign="top">
<h2><span class="refentrytitle"><a name="FsRtpSession.top_of_page"></a>FsRtpSession</span></h2>
<p>FsRtpSession — A RTP session in a <a class="link" href="FsRtpConference.html" title="FsRtpConference"><span class="type">FsRtpConference</span></a></p>
</td>
<td class="gallery_image" valign="top" align="right"></td>
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<div class="refsect1">
<a name="FsRtpSession.properties"></a><h2>Properties</h2>
<div class="informaltable"><table border="0">
<colgroup>
<col width="150px" class="properties_type">
<col width="300px" class="properties_name">
<col width="200px" class="properties_flags">
</colgroup>
<tbody>
<tr>
<td class="property_type"><span class="type">gint</span></td>
<td class="property_name"><a class="link" href="FsRtpSession.html#FsRtpSession--no-rtcp-timeout" title="The “no-rtcp-timeout” property">no-rtcp-timeout</a></td>
<td class="property_flags">Read / Write</td>
</tr>
<tr>
<td class="property_type">
<span class="type">FsRtpHeaderExtensionGList</span> *</td>
<td class="property_name"><a class="link" href="FsRtpSession.html#FsRtpSession--rtp-header-extension-preferences" title="The “rtp-header-extension-preferences” property">rtp-header-extension-preferences</a></td>
<td class="property_flags">Read / Write</td>
</tr>
<tr>
<td class="property_type">
<span class="type">FsRtpHeaderExtensionGList</span> *</td>
<td class="property_name"><a class="link" href="FsRtpSession.html#FsRtpSession--rtp-header-extensions" title="The “rtp-header-extensions” property">rtp-header-extensions</a></td>
<td class="property_flags">Read</td>
</tr>
<tr>
<td class="property_type"><span class="type">guint</span></td>
<td class="property_name"><a class="link" href="FsRtpSession.html#FsRtpSession--send-bitrate" title="The “send-bitrate” property">send-bitrate</a></td>
<td class="property_flags">Read / Write</td>
</tr>
<tr>
<td class="property_type"><span class="type">guint</span></td>
<td class="property_name"><a class="link" href="FsRtpSession.html#FsRtpSession--ssrc" title="The “ssrc” property">ssrc</a></td>
<td class="property_flags">Read / Write</td>
</tr>
<tr>
<td class="property_type">
<span class="type">GObject</span> *</td>
<td class="property_name"><a class="link" href="FsRtpSession.html#FsRtpSession--internal-session" title="The “internal-session” property">internal-session</a></td>
<td class="property_flags">Read</td>
</tr>
</tbody>
</table></div>
</div>
<div class="refsect1">
<a name="FsRtpSession.other"></a><h2>Types and Values</h2>
<div class="informaltable"><table width="100%" border="0">
<colgroup>
<col width="150px" class="name">
<col class="description">
</colgroup>
<tbody><tr>
<td class="datatype_keyword">struct</td>
<td class="function_name"><a class="link" href="FsRtpSession.html#FsRtpSession-struct" title="struct FsRtpSession">FsRtpSession</a></td>
</tr></tbody>
</table></div>
</div>
<div class="refsect1">
<a name="FsRtpSession.object-hierarchy"></a><h2>Object Hierarchy</h2>
<pre class="screen"> GObject
<span class="lineart">╰──</span> FsSession
<span class="lineart">╰──</span> FsRtpSession
</pre>
</div>
<div class="refsect1">
<a name="FsRtpSession.description"></a><h2>Description</h2>
<p>This object represents one session, it is created by called
<code class="function">fs_conference_new_session()</code> on a <a class="link" href="FsRtpConference.html" title="FsRtpConference"><span class="type">FsRtpConference</span></a>. It can be either
Audio or Video. It also represents data send with one and only one
SSRC (although if there is a SSRC collision, that SSRC may change).</p>
<div class="refsect2">
<a name="id-1.2.2.4.6.3"></a><h3>Codec profiles</h3>
<p>
It is possible to define "codec profiles", that is non-autodetected
encoding and decoding pipelines for codecs. It is even possible to declare
entirely new codecs using this method.
To create a profile for a codec, add it to the codec-preferences with
special optional parameters called "farstream-send-profile" and
"farstream-recv-profile", these should contain gst-launch style descriptions
of the encoding or decoding bin.
As a special case, encoding profiles can have more than one unconnected
source pad, all of these pads should produce application/x-rtp of some kind.
The profile will be ignored if not ALL pads match currently negotiated
codecs.
Also, it is possible to declare profiles with only a decoding pipeline,
you will only be able to receive from this codec, the encoding may be a
secondary pad of some other codec.
</p>
</div>
<hr>
<div class="refsect2">
<a name="id-1.2.2.4.6.4"></a><h3>SRTP signature and encryption</h3>
<p>
To tell <a class="link" href="FsRtpSession.html" title="FsRtpSession"><span class="type">FsRtpSession</span></a> to authenticate encrypt the media it is
sending using SRTP, one must set the parameters using a
<span class="type">GstStructure</span> named "FarstreamSRTP" and passing it to
<code class="function">fs_session_set_encryption_parameters()</code>.
The cipher, auth, and key must be specified:
</p>
<div class="refsect3">
<a name="FarstreamSRTP"></a><h4>FarstreamSRTP</h4>
<div class="table"><table>
<tr>
<td><code class="code">"rtp-cipher" and "rtcp-cipher"</code></td>
<td>gchar *</td>
<td>
<p>Encryption algorithm</p>
<p>Possible values: "null", "aes-128-icm" or "aes-256-icm"</p>
</td>
</tr>
<tr>
<td><code class="code">"cipher"</code></td>
<td>gchar *</td>
<td>
<p>Default value for "rtp-cipher" and "rtcp-cipher"</p>
<p>Possible values: "null", "aes-128-icm" or "aes-256-icm"</p>
</td>
</tr>
<tr>
<td><code class="code">"rtp-auth" and "rtcp-auth"</code></td>
<td>gchar *</td>
<td>
<p>Authentication algorithm, can never be null</p>
<p>Possible values: "hmac-sha1-32" or "hmac-sha1-80"</p>
</td>
</tr>
<tr>
<td><code class="code">"auth"</code></td>
<td>gchar *</td>
<td>
<p>Default value for "rtp-auth" and "rtcp-auth"</p>
<p>Possible values: "hmac-sha1-32" or "hmac-sha1-80"</p>
</td>
</tr>
<tr>
<td><code class="code">"key"</code></td>
<td><span class="type">GstBuffer</span></td>
<td>Size must be 30 if cipher is "aes-128-icm" and 46 if cipher is
"aes-256-icm" </td>
</tr>
</table></div>
</div>
<p>
</p>
</div>
</div>
<div class="refsect1">
<a name="FsRtpSession.functions_details"></a><h2>Functions</h2>
<p></p>
</div>
<div class="refsect1">
<a name="FsRtpSession.other_details"></a><h2>Types and Values</h2>
<div class="refsect2">
<a name="FsRtpSession-struct"></a><h3>struct FsRtpSession</h3>
<pre class="programlisting">struct FsRtpSession;</pre>
</div>
</div>
<div class="refsect1">
<a name="FsRtpSession.property-details"></a><h2>Property Details</h2>
<div class="refsect2">
<a name="FsRtpSession--no-rtcp-timeout"></a><h3>The <code class="literal">“no-rtcp-timeout”</code> property</h3>
<pre class="programlisting"> “no-rtcp-timeout” <span class="type">gint</span></pre>
<p>This is the time (in ms) after which data received without RTCP is attached the FsStream, this only works if there is only one FsStream. -1 will wait forever. 0 will not wait for RTCP and attach it immediataly to the FsStream and prohibit the creation of a second FsStream.</p>
<p>Flags: Read / Write</p>
<p>Allowed values: >= G_MAXULONG</p>
<p>Default value: 7000</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--rtp-header-extension-preferences"></a><h3>The <code class="literal">“rtp-header-extension-preferences”</code> property</h3>
<pre class="programlisting"> “rtp-header-extension-preferences” <span class="type">FsRtpHeaderExtensionGList</span> *</pre>
<p>GList of RTP Header extensions that are locally supported and desired by the application.</p>
<p>Flags: Read / Write</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--rtp-header-extensions"></a><h3>The <code class="literal">“rtp-header-extensions”</code> property</h3>
<pre class="programlisting"> “rtp-header-extensions” <span class="type">FsRtpHeaderExtensionGList</span> *</pre>
<p>GList of RTP Header extensions that have been negotiated and will be used when sending of receiving RTP packets.</p>
<p>Flags: Read</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--send-bitrate"></a><h3>The <code class="literal">“send-bitrate”</code> property</h3>
<pre class="programlisting"> “send-bitrate” <span class="type">guint</span></pre>
<p>The bitrate that the session will try to send at in bits/sec.</p>
<p>Flags: Read / Write</p>
<p>Default value: 0</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--ssrc"></a><h3>The <code class="literal">“ssrc”</code> property</h3>
<pre class="programlisting"> “ssrc” <span class="type">guint</span></pre>
<p>This is the current SSRC used to send data (defaults to a random value).</p>
<p>Flags: Read / Write</p>
<p>Default value: 0</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--internal-session"></a><h3>The <code class="literal">“internal-session”</code> property</h3>
<pre class="programlisting"> “internal-session” <span class="type">GObject</span> *</pre>
<p>Internal RTPSession object from rtpbin.</p>
<p>Flags: Read</p>
</div>
</div>
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