<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><span style="font-size: 14px;">Hi Sebastian,</span><div><span style="font-size: 14px;">digging with the issue I see a compatibility issue with the soft phone I’m using(Linphphone), it seems its vp8 decoder with name "WebM Project VP8 Decoder v0.9.7-pl” is not compatible with your encoder (I got gstreamer-1.2.pkg, you provide us)</span></div><div><span style="font-size: 14px;"><br></span></div><div><span style="font-size: 14px;">And the only information I got was:</span></div><div><span style="background-color: rgb(255, 255, 255); color: rgb(34, 34, 34); font-family: 'courier new', monospace; font-size: 14px;">vpx_codec_decode failed : 5 Bitstream not supported by this decoder ((null))</span></div><div><span style="font-size: 14px;"><br></span></div><div><span style="font-size: 14px;">People develop webrtc2sip server were having the same issue, I’ve found some threads about this issue</span></div><div><ul class="MailOutline"><li><span style="font-size: 14px;"><a href="https://groups.google.com/forum/#!msg/discuss-webrtc/xCxqzxopsDM/jIKtkA_StOgJ">https://groups.google.com/forum/#!msg/discuss-webrtc/xCxqzxopsDM/jIKtkA_StOgJ</a></span></li><li><span style="font-size: 14px;"><a href="https://groups.google.com/forum/#!topic/doubango/OH-nOn5w-ng">https://groups.google.com/forum/#!topic/doubango/OH-nOn5w-ng</a></span></li></ul><div><span style="font-size: 14px;">They solved disabling extension in vp8 encoding #define <span style="white-space: pre-wrap;">TDAV_VP8_DISABLE_EXTENSION in</span></span></div><ul class="MailOutline"><li><span style="font-size: 14px;"><a href="http://doubango.googlecode.com/svn-history/r796/branches/2.0/doubango/tinyDAV/src/codecs/vpx/tdav_codec_vp8.c">http://doubango.googlecode.com/svn-history/r796/branches/2.0/doubango/tinyDAV/src/codecs/vpx/tdav_codec_vp8.c</a> </span></li></ul></div><div><span style="font-size: 14px;"><br></span></div><div><span style="font-size: 14px;"><br></span></div><div><div><span style="font-size: 14px;"><span style="background-color: rgb(255, 255, 255);"><font color="#222222">This is the header of the first packet I sent to the </font></span><font color="#222222">softphone</font></span></div><div><span style="font-size: 14px;">90:80:7c:10:32:00:9d:01:2a:a0:00:78:00:00:47:08:85:85:88:85:84:88…</span></div><div><span style="font-size: 14px;"><br></span></div><div><span style="font-size: 14px;">If I’ve understood correctly these discussion, the header 90:80 is not compliant with WebM Vp8 decoder used linphone, because it cannot interpret such header.</span></div></div><div><span style="font-size: 14px;"><br></span></div><div><span style="font-size: 14px;">What do you think? Have you experience on these compatibility issues?</span></div><div><span style="font-size: 14px;"><br></span></div><div><span style="font-size: 14px;">Thanks</span></div><div><span style="font-size: 14px;">Elio</span></div><div><span style="font-size: 14px;"><br></span></div><div><br></div><div><span style="color: rgb(209, 47, 27); font-family: Menlo; font-size: 11px;">On 31 Oct 2013, at 12:20, Sebastian Dröge <<a href="mailto:sebastian@centricular.com">sebastian@centricular.com</a>> wrote:</span></div><div><div><br class="Apple-interchange-newline"><blockquote type="cite">On Do, 2013-10-31 at 00:18 +0100, Elio Francesconi wrote:<br><blockquote type="cite">Great! it is working thank you<br>I’m wondering if you can help me also with this issue:<br><br>I’ve created a sample application with Gstreamer 1.x to open an rtp session, it can be summarised with this syntax:<br><br>gst-launch -v videotestsrc ! "video/x-raw, width=640, height=480, format=AYUV” ! vp8enc ! rtpvp8pay ! udpsink host=127.0.0.1 port=5011<br><br>On the client I don’t see at all video (black screen)<br>Do you see something wrong in this pipeline?<br>Can be the format I’m using wrong?<br></blockquote><br>vp8enc only supports I420, not AYUV. That might be one problem :)<br><br>On the receiver side, first of all check if you can receive any RTP<br>packets at all with something like<br>gst-launch-1.0 -v udpsrc multicast-group=127.0.0.1 port=5011 ! fakesink silent=false<br><br>This should tell you if packets arrive. If they don't there's something<br>wrong on the sender already. If they do iteratively build the pipeline<br>to check from which point things go wrong.<br><br>-- <br>Sebastian Dröge <<a href="mailto:sebastian@centricular.com">sebastian@centricular.com</a>><br>Centricular Ltd - <a href="http://www.centricular.com">http://www.centricular.com</a><br>Expertise, Straight from the Source<br>_______________________________________________<br>gstreamer-devel mailing list<br><a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a><br>http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel<br></blockquote></div><br></div></body></html>