RTP-audio-video

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RTP-audio-video

vaisakhn7
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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***** Confidentiality Statement/Disclaimer *****  

This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt.

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.


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Re: RTP-audio-video

AJAY GAUTAM
Just try :
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=TRUE
video is working!! not able hear sound !!!


On Wed, Apr 1, 2009 at 2:14 PM, vaisakh.n <[hidden email]> wrote:
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

***** Confidentiality Statement/Disclaimer *****

This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt.
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.
***** Confidentiality Statement/Disclaimer *****

This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt.

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.


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--
Thanx & Regards
Ajay Gautam
+91-9717785580

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