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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">Dear GST Community,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">i have implemented a rtsp server with the gst-rtsp-server module. The server receives an audio signal via an interaudiosrc and a video signal via an intervideosrc:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">
<i>intervideosrc channel=videosrc </i></span><i><span style="font-size:10.0pt;font-family:Wingdings">à</span></i><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> videoconvert
</span></i><i><span style="font-size:10.0pt;font-family:Wingdings">à</span></i><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> vaapih264enc bitrate=5000 zerolatency=true
</span></i><i><span style="font-size:10.0pt;font-family:Wingdings">à</span></i><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> rtph264pay pt=96 name=pay0<o:p></o:p></span></i></p>
<p class="MsoNormal"><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> interaudiosrc channel=audiosrc
</span></i><i><span style="font-size:10.0pt;font-family:Wingdings">à</span></i><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> opusenc bitrate=64
</span></i><i><span style="font-size:10.0pt;font-family:Wingdings">à</span></i><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> queue
</span></i><i><span style="font-size:10.0pt;font-family:Wingdings">à</span></i><i><span style="font-size:10.0pt;font-family:"Arial",sans-serif"> rtpopuspay pt=97 name=pay1<o:p></o:p></span></i></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">When I try to view the stream via an rtsp client, for example VLC Player the stream works, but the audio signal is about 1 second behind the video signal. I already tried different
audio encoders and different queue parameters like buffer size and leaky queues, but nothing solved the problem.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">When I start the same pipeline as above but mux both signals and write them into a filesink they are synchronized.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">I know that the server module internally uses the multiudpsink element. Is there any possibility to edit multiudpsink parameters from outside?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">Does anybody have any ideas about this problem? I already read every google result I could find about it.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">Best Regards,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">Felix Sternkopf<o:p></o:p></span></p>
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