gstreamer/docs/plugins/inspect/plugin-gstrtpmanager.xml

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<plugin>
<name>gstrtpmanager</name>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
<version>0.10.13.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins git/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>gstrtpbin</name>
<longname>RTP Bin</longname>
<class>Filter/Network/RTP</class>
<description>Implement an RTP bin</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
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<name>recv_rtcp_sink_%d</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtcp</details>
</caps>
<caps>
<name>recv_rtp_sink_%d</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtp</details>
</caps>
<caps>
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<name>send_rtp_sink_%d</name>
<direction>sink</direction>
<presence>request</presence>
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<details>application/x-rtp</details>
</caps>
<caps>
<name>recv_rtp_src_%d_%d_%d</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtp</details>
</caps>
<caps>
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<name>send_rtp_src_%d</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtp</details>
</caps>
<caps>
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<name>send_rtcp_src_%d</name>
<direction>source</direction>
<presence>request</presence>
<details>application/x-rtcp</details>
</caps>
</pads>
</element>
<element>
<name>gstrtpjitterbuffer</name>
<longname>RTP packet jitter-buffer</longname>
<class>Filter/Network/RTP</class>
<description>A buffer that deals with network jitter and other transmission faults</description>
<author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;, Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtp, clock-rate=(int)[ 1, 2147483647 ]</details>
</caps>
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<caps>
<name>sink_rtcp</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtcp</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp</details>
</caps>
</pads>
</element>
<element>
<name>gstrtpptdemux</name>
<longname>RTP Demux</longname>
<class>Demux/Network/RTP</class>
<description>Parses codec streams transmitted in the same RTP session</description>
<author>Kai Vehmanen &lt;kai.vehmanen@nokia.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtp</details>
</caps>
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<caps>
<name>src_%d</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtp, payload=(int)[ 0, 255 ]</details>
</caps>
</pads>
</element>
<element>
<name>gstrtpsession</name>
<longname>RTP Session</longname>
<class>Filter/Network/RTP</class>
<description>Implement an RTP session</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
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<name>recv_rtcp_sink</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtcp</details>
</caps>
<caps>
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<name>recv_rtp_sink</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtp</details>
</caps>
<caps>
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<name>send_rtp_sink</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtp</details>
</caps>
<caps>
<name>recv_rtp_src</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtp</details>
</caps>
<caps>
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<name>send_rtp_src</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtp</details>
</caps>
<caps>
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<name>sync_src</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtcp</details>
</caps>
<caps>
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<name>send_rtcp_src</name>
<direction>source</direction>
<presence>request</presence>
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<details>application/x-rtcp</details>
</caps>
</pads>
</element>
<element>
<name>gstrtpssrcdemux</name>
<longname>RTP SSRC Demux</longname>
<class>Demux/Network/RTP</class>
<description>Splits RTP streams based on the SSRC</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
<name>rtcp_sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtcp</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtp</details>
</caps>
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<caps>
<name>rtcp_src_%d</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtcp</details>
</caps>
<caps>
<name>src_%d</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>application/x-rtp</details>
</caps>
</pads>
</element>
</elements>
</plugin>