[gst-devel] RTP-audio-video

[vaisakh.n]

Hi,
I was trying to implement videoconferencing through gstreamer
for that
In sender side i ran the pipeline like this
/*****************/
./gst-launch -v v4lsrc name=source  ! tee name=t  t. ! queue ! 
ffenc_h263p ! rtph263ppay ! udpsink host=10.1.11.33 port=5000  t. ! 
queue ! ffmpegcolorspace !  sdlvideosink    osssrc ! 
audio/x-raw-int,rate=8000,channels=1,depth=16  !  audioconvert ! 
vorbisenc  ! rtpvorbispay ! udpsink host=10.1.11.33 port=6000

/**************************/
In receiver side  pipeline is ./gst-launch-0.10 udpsrc   port=5000 
caps="application/x-rtp, media=(string)video,clock-rate=(int)90000, 
encoding-name=(string)H263-1998" ! rtph263pdepay ! ffdec_h263 ! 
sdlvideosink sync=false { udpsrc  port=6000  caps="application/x-rtp, 
media=(string)audio, clock-rate=(int)8000, encoding-name=(string)VORBIS, 
encoding-params=(string)1, configuration=(string)\" STRING"  !   
rtpvorbisdepay ! vorbisdec ! audioconvert ! osssink  sync=false
video is working!! not able hear sound !!!

AND  IN RECEIVER SIDE
/***********************/
dropped 64 samples
/GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = "encoding 
at quality level 0.30"
WARNING: from element /GstPipeline:pipeline0/GstOssSrc:osssrc0: Can't 
record audio fast enough
Additional debug info:
gstbaseaudiosrc.c(806): gst_base_audio_src_create (): 
/GstPipeline:pipeline0/GstOssSrc:osssrc0:
dropped 256 samples
 /GstPipeline:pipeline0/GstVorbisEnc:vorbisenc0: last-message = 
"encoding at quality level 0.30"
/**************************/
Getting this warning continuously  and sound is full of disturbance

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The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message.