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<div class="refentry">
<a name="gst-plugins-good-plugins-rtprtxreceive"></a><div class="titlepage"></div>
<div class="refnamediv"><table width="100%"><tr>
<td valign="top">
<h2><span class="refentrytitle"><a name="gst-plugins-good-plugins-rtprtxreceive.top_of_page"></a>rtprtxreceive</span></h2>
<p>rtprtxreceive</p>
</td>
<td class="gallery_image" valign="top" align="right"></td>
</tr></table></div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.properties"></a><h2>Properties</h2>
<div class="informaltable"><table class="informaltable" 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"><a href="../glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive--num-rtx-requests" title="The “num-rtx-requests” property">num-rtx-requests</a></td>
<td class="property_flags">Read</td>
</tr>
<tr>
<td class="property_type"><a href="../glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive--num-rtx-packets" title="The “num-rtx-packets” property">num-rtx-packets</a></td>
<td class="property_flags">Read</td>
</tr>
<tr>
<td class="property_type"><a href="../glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive--num-rtx-assoc-packets" title="The “num-rtx-assoc-packets” property">num-rtx-assoc-packets</a></td>
<td class="property_flags">Read</td>
</tr>
<tr>
<td class="property_type">
<a href="../gstreamer-1.0/GstStructure.html#GstStructure-struct"><span class="type">GstStructure</span></a> *</td>
<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive--payload-type-map" title="The “payload-type-map” property">payload-type-map</a></td>
<td class="property_flags">Read / Write</td>
</tr>
</tbody>
</table></div>
</div>
<a name="GstRtpRtxReceive"></a><div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.other"></a><h2>Types and Values</h2>
<div class="informaltable"><table class="informaltable" 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="gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive-struct" title="struct GstRtpRtxReceive">GstRtpRtxReceive</a></td>
</tr></tbody>
</table></div>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.object-hierarchy"></a><h2>Object Hierarchy</h2>
<pre class="screen"> <a href="../gobject/gobject-The-Base-Object-Type.html#GObject-struct">GObject</a>
<span class="lineart">╰──</span> <a href="../gobject/gobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a>
<span class="lineart">╰──</span> <a href="../gstreamer-1.0/GstObject.html#GstObject-struct">GstObject</a>
<span class="lineart">╰──</span> <a href="../gstreamer-1.0/GstElement.html#GstElement-struct">GstElement</a>
<span class="lineart">╰──</span> GstRtpRtxReceive
</pre>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.description"></a><h2>Description</h2>
<p>rtprtxreceive listens to the retransmission events from the
downstream rtpjitterbuffer and remembers the SSRC (ssrc1) of the stream and
the sequence number that was requested. When it receives a packet with
a sequence number equal to one of the ones stored and with a different SSRC,
it identifies the new SSRC (ssrc2) as the retransmission stream of ssrc1.
From this point on, it replaces ssrc2 with ssrc1 in all packets of the
ssrc2 stream and flags them as retransmissions, so that rtpjitterbuffer
can reconstruct the original stream.</p>
<p>This algorithm is implemented as specified in RFC 4588.</p>
<p>This element is meant to be used with rtprtxsend on the sender side.
See <a class="link" href="gst-plugins-good-plugins-rtprtxsend.html#GstRtpRtxSend"><span class="type">GstRtpRtxSend</span></a></p>
<p>Below you can see some examples that illustrate how rtprtxreceive and
rtprtxsend fit among the other rtp elements and how they work internally.
Normally, hoewever, you should avoid using such pipelines and use
rtpbin instead, with its <a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-request-aux-sender" title="The “request-aux-sender” signal"><span class="type">“request-aux-sender”</span></a> and
<a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-request-aux-receiver" title="The “request-aux-receiver” signal"><span class="type">“request-aux-receiver”</span></a> signals. See <a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin"><span class="type">GstRtpBin</span></a>.</p>
<div class="refsect2">
<a name="id-1.2.211.7.6"></a><h3>Example pipelines</h3>
<div class="informalexample">
<table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1
2
3
4
5
6
7
8
9</pre></td>
<td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
audiotestsrc is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">96</span> <span class="gtkdoc opt">!</span> \
rtprtxsend payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp-pt-map,96=(uint)97"</span> <span class="gtkdoc opt">!</span> \
rtpsession<span class="gtkdoc opt">.</span>send_rtp_sink \
rtpsession<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.01</span> <span class="gtkdoc opt">!</span> \
udpsink host<span class="gtkdoc opt">=</span><span class="string">"127.0.0.1"</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">"127.0.0.1"</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> \
sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
</tr>
</tbody>
</table>
</div>
<p> Send audio stream through port 5000 (5001 and 5002 are just the rtcp
link with the receiver)</p>
<div class="informalexample">
<table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1
2
3
4
5
6
7
8
9
10</pre></td>
<td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96"</span> <span class="gtkdoc opt">!</span> \
rtpsession<span class="gtkdoc opt">.</span>recv_rtp_sink \
rtpsession<span class="gtkdoc opt">.</span>recv_rtp_src <span class="gtkdoc opt">!</span> \
rtprtxreceive payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp-pt-map,96=(uint)97"</span> <span class="gtkdoc opt">!</span> \
rtpssrcdemux <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> \
rtpopusdepay <span class="gtkdoc opt">!</span> opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioresample <span class="gtkdoc opt">!</span> autoaudiosink \
rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> \
udpsink host<span class="gtkdoc opt">=</span><span class="string">"127.0.0.1"</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span> \
udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink</pre></td>
</tr>
</tbody>
</table>
</div>
<p> Receive audio stream from port 5000 (5001 and 5002 are just the rtcp
link with the sender)</p>
<p>In this example we can see a simple streaming of an OPUS stream with some
of the packets being artificially dropped by the identity element.
Thanks to retransmission, you should still hear a clear sound when setting
drop-probability to something greater than 0.</p>
<p>Internally, the rtpjitterbuffer will generate a custom upstream event,
GstRTPRetransmissionRequest, when it detects that one packet is missing.
Then this request is translated to a FB NACK in the rtcp link by rtpsession.
Finally the rtpsession of the sender side will re-convert it in a
GstRTPRetransmissionRequest that will be handled by rtprtxsend. rtprtxsend
will then re-send the missing packet with a new srrc and a different payload
type (here, 97), but with the same original sequence number. On the receiver
side, rtprtxreceive will associate this new stream with the original and
forward the retransmission packets to rtpjitterbuffer with the original
ssrc and payload type.</p>
<div class="informalexample">
<table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1
2
3
4
5
6
7
8
9
10
11
12
13</pre></td>
<td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
audiotestsrc is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">97</span> seqnum<span class="gtkdoc opt">-</span>offset<span class="gtkdoc opt">=</span><span class="number">1</span> <span class="gtkdoc opt">!</span> \
rtprtxsend payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp-pt-map,97=(uint)99"</span> <span class="gtkdoc opt">!</span> \
funnel name<span class="gtkdoc opt">=</span>f <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>send_rtp_sink \
audiotestsrc freq<span class="gtkdoc opt">=</span><span class="number">660.0</span> is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> \
rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">97</span> seqnum<span class="gtkdoc opt">-</span>offset<span class="gtkdoc opt">=</span><span class="number">100</span> <span class="gtkdoc opt">!</span> \
rtprtxsend payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp-pt-map,97=(uint)99"</span> <span class="gtkdoc opt">!</span> \
f<span class="gtkdoc opt">.</span> \
rtpsession<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.01</span> <span class="gtkdoc opt">!</span> \
udpsink host<span class="gtkdoc opt">=</span><span class="string">"127.0.0.1"</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">"127.0.0.1"</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> \
sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
</tr>
</tbody>
</table>
</div>
<p> Send two audio streams to port 5000.</p>
<div class="informalexample">
<table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
<tbody>
<tr>
<td class="listing_lines" align="right"><pre>1
2
3
4
5
6
7
8
9
10
11
12
13</pre></td>
<td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97"</span> <span class="gtkdoc opt">!</span> \
rtpsession<span class="gtkdoc opt">.</span>recv_rtp_sink \
rtpsession<span class="gtkdoc opt">.</span>recv_rtp_src <span class="gtkdoc opt">!</span> \
rtprtxreceive payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">"application/x-rtp-pt-map,97=(uint)99"</span> <span class="gtkdoc opt">!</span> \
rtpssrcdemux name<span class="gtkdoc opt">=</span>demux \
demux<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpopusdepay <span class="gtkdoc opt">!</span> \
opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
demux<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpopusdepay <span class="gtkdoc opt">!</span> \
opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">"127.0.0.1"</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> \
sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
</tr>
</tbody>
</table>
</div>
<p> Receive two audio streams from port 5000.</p>
<p>In this example we are streaming two streams of the same type through the
same port. They, however, are using a different SSRC (ssrc is randomly
generated on each payloader - rtpopuspay in this example), so they can be
identified and demultiplexed by rtpssrcdemux on the receiver side. This is
an example of SSRC-multiplexing.</p>
<p>It is important here to use a different starting sequence number
(seqnum-offset), since this is the only means of identification that
rtprtxreceive uses the very first time to identify retransmission streams.
It is an error, according to RFC4588 to have two retransmission requests for
packets belonging to two different streams but with the same sequence number.
Note that the default seqnum-offset value (-1, which means random) would
work just fine, but it is overriden here for illustration purposes.</p>
</div>
<div class="refsynopsisdiv">
<h2>Synopsis</h2>
<div class="refsect2">
<a name="id-1.2.211.7.7.1"></a><h3>Element Information</h3>
<div class="variablelist"><table border="0" class="variablelist">
<colgroup>
<col align="left" valign="top">
<col>
</colgroup>
<tbody>
<tr>
<td><p><span class="term">plugin</span></p></td>
<td>
<a class="link" href="gst-plugins-good-plugins-plugin-rtpmanager.html#plugin-rtpmanager">rtpmanager</a>
</td>
</tr>
<tr>
<td><p><span class="term">author</span></p></td>
<td>Julien Isorce <julien.isorce@collabora.co.uk></td>
</tr>
<tr>
<td><p><span class="term">class</span></p></td>
<td>Codec</td>
</tr>
</tbody>
</table></div>
</div>
<hr>
<div class="refsect2">
<a name="id-1.2.211.7.7.2"></a><h3>Element Pads</h3>
<div class="variablelist"><table border="0" class="variablelist">
<colgroup>
<col align="left" valign="top">
<col>
</colgroup>
<tbody>
<tr>
<td><p><span class="term">name</span></p></td>
<td>sink</td>
</tr>
<tr>
<td><p><span class="term">direction</span></p></td>
<td>sink</td>
</tr>
<tr>
<td><p><span class="term">presence</span></p></td>
<td>always</td>
</tr>
<tr>
<td><p><span class="term">details</span></p></td>
<td>application/x-rtp</td>
</tr>
</tbody>
</table></div>
<div class="variablelist"><table border="0" class="variablelist">
<colgroup>
<col align="left" valign="top">
<col>
</colgroup>
<tbody>
<tr>
<td><p><span class="term">name</span></p></td>
<td>src</td>
</tr>
<tr>
<td><p><span class="term">direction</span></p></td>
<td>source</td>
</tr>
<tr>
<td><p><span class="term">presence</span></p></td>
<td>always</td>
</tr>
<tr>
<td><p><span class="term">details</span></p></td>
<td>application/x-rtp</td>
</tr>
</tbody>
</table></div>
</div>
</div>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.functions_details"></a><h2>Functions</h2>
<p></p>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.other_details"></a><h2>Types and Values</h2>
<div class="refsect2">
<a name="GstRtpRtxReceive-struct"></a><h3>struct GstRtpRtxReceive</h3>
<pre class="programlisting">struct GstRtpRtxReceive;</pre>
</div>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.property-details"></a><h2>Property Details</h2>
<div class="refsect2">
<a name="GstRtpRtxReceive--num-rtx-requests"></a><h3>The <code class="literal">“num-rtx-requests”</code> property</h3>
<pre class="programlisting"> “num-rtx-requests” <a href="../glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
<p>Number of retransmission events received.</p>
<p>Flags: Read</p>
<p>Default value: 0</p>
</div>
<hr>
<div class="refsect2">
<a name="GstRtpRtxReceive--num-rtx-packets"></a><h3>The <code class="literal">“num-rtx-packets”</code> property</h3>
<pre class="programlisting"> “num-rtx-packets” <a href="../glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
<p> Number of retransmission packets received.</p>
<p>Flags: Read</p>
<p>Default value: 0</p>
</div>
<hr>
<div class="refsect2">
<a name="GstRtpRtxReceive--num-rtx-assoc-packets"></a><h3>The <code class="literal">“num-rtx-assoc-packets”</code> property</h3>
<pre class="programlisting"> “num-rtx-assoc-packets” <a href="../glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
<p>Number of retransmission packets correctly associated with retransmission requests.</p>
<p>Flags: Read</p>
<p>Default value: 0</p>
</div>
<hr>
<div class="refsect2">
<a name="GstRtpRtxReceive--payload-type-map"></a><h3>The <code class="literal">“payload-type-map”</code> property</h3>
<pre class="programlisting"> “payload-type-map” <a href="../gstreamer-1.0/GstStructure.html#GstStructure-struct"><span class="type">GstStructure</span></a> *</pre>
<p>Map of original payload types to their retransmission payload types.</p>
<p>Flags: Read / Write</p>
</div>
</div>
<div class="refsect1">
<a name="gst-plugins-good-plugins-rtprtxreceive.see-also"></a><h2>See Also</h2>
<p>rtprtxsend, rtpsession, rtpjitterbuffer</p>
</div>
</div>
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