<div dir="ltr">Hi,<div><br></div><div>Thanks for your reply.</div><div>As you suggested I tried using rtprtxqueue on receiver side(pipeline below):</div><div><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div>
<div><span style="font-family:arial,sans-serif;font-size:13px">gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000</span><br style="font-family:arial,sans-serif;font-size:13px"><span style="font-family:arial,sans-serif;font-size:13px">caps="application/x-rtp, media=audio, clock-rate=8000, encoding-name=PCMA,</span><br style="font-family:arial,sans-serif;font-size:13px">
<span style="font-family:arial,sans-serif;font-size:13px">payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! <b>rtprtxqueue</b> ! rtppcmadepay ! alawdec !</span><br style="font-family:arial,sans-serif;font-size:13px"><span style="font-family:arial,sans-serif;font-size:13px">alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0</span><br style="font-family:arial,sans-serif;font-size:13px">
<span style="font-family:arial,sans-serif;font-size:13px">! udpsink host=192.154.10.21 port=5005 sync=false async=false</span><br style="font-family:arial,sans-serif;font-size:13px"><span style="font-family:arial,sans-serif;font-size:13px">--gst-debug=3</span><br>
</div><div><br></div><div>Still I am not able to see any RTCP packets other than reports from receiver to sender in wireshark.</div><div>Is above pipeline correct?</div><div>As per my understanding there should be RTCP request from receiver to sender to send again lost packets.</div>
<div>Is this correct?</div><div><br></div><div>Thanks,</div><div>Abhilash.</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Dec 11, 2013 at 3:21 PM, George Kiagiadakis [via GStreamer-devel] <span dir="ltr"><<a href="/user/SendEmail.jtp?type=node&node=4663967&i=0" target="_top" rel="nofollow" link="external">[hidden email]</a>></span> wrote:<br>
<blockquote style='border-left:2px solid #CCCCCC;padding:0 1em' class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">
On Wed, Dec 11, 2013 at 10:29 AM, abhilash <<a href="http://user/SendEmail.jtp?type=node&node=4663966&i=0" rel="nofollow" link="external" target="_blank">[hidden email]</a>> wrote:
<div><div class='shrinkable-quote'><br>> Hi,
<br>>
<br>> I am trying out a client-server based audio streaming application using
<br>> gstreamer 1.2.0 version, running on 2 different Linux machines on network. I
<br>> want to check the functionality of retransmissions as mentioned in the
<br>> "rtpbin" element and I tried setting the property "do-retransmissions=true".
<br>> I am simulating packet loss, packet delay & reorder on the server side. The
<br>> observation is as follows -
<br>> 1. I am able to see RTCP Sender and Receiver reports being sent between
<br>> server and client on Wireshark.
<br>> 2. I am able to see dropped RTP packet sequence numbers in Wireshark trace,
<br>> however I am not able to see any RTP retransmissions event going from client
<br>> to server.
<br>>
<br>> The pipelines at the server and client side are as mentioned below -
<br>>
<br>> Server:
<br>> gst-launch-1.0 -v rtpbin name=rtpbin do-retransmission=true audiotestsrc
<br>> freq=1000 ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0
<br>> rtpbin.send_rtp_src_0 ! udpsink host=192.154.10.20 port=5000
<br>> rtpbin.send_rtcp_src_0 ! udpsink host=192.154.10.20 port=5001 sync=false
<br>> async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 --gst-debug=3
<br>>
<br>> Client:
<br>> gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
<br>> caps="application/x-rtp, media=audio, clock-rate=8000, encoding-name=PCMA,
<br>> payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtppcmadepay ! alawdec !
<br>> alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0
<br>> ! udpsink host=192.154.10.21 port=5005 sync=false async=false
<br>> --gst-debug=3
<br>>
<br>> Can anyone please let me know if anything is missed out above and why I
<br>> am not seeing any RTP retransmissions ?
<br>>
</div></div></div></div><div class="im">Hi,
<br><br>do-retransmission=true should be set on the receiver (client) rtpbin.
<br>On the sender it has no effect. Also, since you are on 1.2, you need
<br>to add rtprtxqueue in your pipeline after the payloader. Note though
<br>that this does not implement RFC4588-compliant retransmission packets.
<br>An RFC4588-compliant implementation is under review in bugzilla at the
<br>moment.
<br><br>Regards,
<br>George
<br>_______________________________________________
<br>gstreamer-devel mailing list
<br></div><a href="http://user/SendEmail.jtp?type=node&node=4663966&i=1" rel="nofollow" link="external" target="_blank">[hidden email]</a>
<br><a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="nofollow" link="external" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><br>
<br>
<br>
<hr noshade size="1" color="#cccccc">
<div style="color:#444;font:12px tahoma,geneva,helvetica,arial,sans-serif">
<div style="font-weight:bold">If you reply to this email, your message will be added to the discussion below:</div>
<a href="http://gstreamer-devel.966125.n4.nabble.com/RTP-retransmission-mechanism-in-gstreamer-1-2-0-tp4663965p4663966.html" target="_blank" rel="nofollow" link="external">http://gstreamer-devel.966125.n4.nabble.com/RTP-retransmission-mechanism-in-gstreamer-1-2-0-tp4663965p4663966.html</a>
</div>
<div style="color:#666;font:11px tahoma,geneva,helvetica,arial,sans-serif;margin-top:.4em;line-height:1.5em">
To unsubscribe from RTP retransmission mechanism in gstreamer 1.2.0, <a href="" target="_blank" rel="nofollow" link="external">click here</a>.<br>
<a href="http://gstreamer-devel.966125.n4.nabble.com/template/NamlServlet.jtp?macro=macro_viewer&id=instant_html%21nabble%3Aemail.naml&base=nabble.naml.namespaces.BasicNamespace-nabble.view.web.template.NabbleNamespace-nabble.view.web.template.NodeNamespace&breadcrumbs=notify_subscribers%21nabble%3Aemail.naml-instant_emails%21nabble%3Aemail.naml-send_instant_email%21nabble%3Aemail.naml" rel="nofollow" style="font:9px serif" target="_blank" link="external">NAML</a>
</div></blockquote></div><br></div>
<br/><hr align="left" width="300" />
View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/RTP-retransmission-mechanism-in-gstreamer-1-2-0-tp4663965p4663967.html">Re: RTP retransmission mechanism in gstreamer 1.2.0</a><br/>
Sent from the <a href="http://gstreamer-devel.966125.n4.nabble.com/">GStreamer-devel mailing list archive</a> at Nabble.com.<br/>