RTP stream: change the payload type

[Elio Francesconi]

Hi Sebastian,
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)

And the only information I got was:
vpx_codec_decode failed : 5 Bitstream not supported by this decoder ((null))

People develop webrtc2sip server were having the same issue, I’ve found some threads about this issue
https://groups.google.com/forum/#!msg/discuss-webrtc/xCxqzxopsDM/jIKtkA_StOgJ
https://groups.google.com/forum/#!topic/doubango/OH-nOn5w-ng
They solved disabling extension in vp8 encoding #define TDAV_VP8_DISABLE_EXTENSION in
http://doubango.googlecode.com/svn-history/r796/branches/2.0/doubango/tinyDAV/src/codecs/vpx/tdav_codec_vp8.c 


This is the header of the first packet I sent to the softphone
90:80:7c:10:32:00:9d:01:2a:a0:00:78:00:00:47:08:85:85:88:85:84:88…

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.

What do you think? Have you experience on these compatibility issues?

Thanks
Elio


On 31 Oct 2013, at 12:20, Sebastian Dröge <sebastian at centricular.com> wrote:

> On Do, 2013-10-31 at 00:18 +0100, Elio Francesconi wrote:
>> Great! it is working thank you
>> I’m wondering if you can help me also with this issue:
>> 
>> I’ve created a sample application with Gstreamer 1.x to open an rtp session, it can be summarised with this syntax:
>> 
>> gst-launch -v videotestsrc ! "video/x-raw, width=640, height=480, format=AYUV” ! vp8enc ! rtpvp8pay ! udpsink host=127.0.0.1 port=5011
>> 
>> On the client I don’t see at all video (black screen)
>> Do you see something wrong in this pipeline?
>> Can be the format I’m using wrong?
> 
> vp8enc only supports I420, not AYUV. That might be one problem :)
> 
> On the receiver side, first of all check if you can receive any RTP
> packets at all with something like
> gst-launch-1.0 -v udpsrc multicast-group=127.0.0.1 port=5011 ! fakesink silent=false
> 
> This should tell you if packets arrive. If they don't there's something
> wrong on the sender already. If they do iteratively build the pipeline
> to check from which point things go wrong.
> 
> -- 
> Sebastian Dröge <sebastian at centricular.com>
> Centricular Ltd - http://www.centricular.com
> Expertise, Straight from the Source
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> gstreamer-devel at lists.freedesktop.org
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