Sending and receiving RTP audio

Kapil Agrawal kapil.agl at gmail.com
Thu Sep 8 02:48:04 PDT 2011


Mat,

I think this is related to the data being lost during transmission, as I
faced similar issue in one project.
I fixed that by configuring rtppay in such a way that mtu is fixed and of
some 1400 bytes.

Not sure if you have similar issue.

Best Luck
Kapil

On Thu, Sep 8, 2011 at 3:04 PM, Matthias Dodt <MDodt at xion-medical.com>wrote:

> Hey guys!
>
> I use OSSBuild with Gstreamer 10.7 (Beta4) on Win7 to transmit an audio
> test signal via UDP/Multicast. It works, but the sound is awful on the
> receiver side.
>
> Sender:
> gst-launch -v gstrtpbin name=rtpbin audiotestsrc ! audioconvert !
> rtpL16pay ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink
> port=5002 host=224.0.1.5  rtpbin.send_rtcp_src_1 ! udpsink port=5003
> host=224.0.1.5 sync=false async=false udpsrc port=5007 !
> rtpbin.recv_rtcp_sink_1
>
> Receiver:
> gst-launch -v gstrtpbin name=rtpbin udpsrc uri=udp://224.0.1.5:5002
> caps=application/x-rtp,clock-rate=(int)44100,encoding-name=(string)L16,m
> edia=(string)audio,channels=(int)1 ! rtpbin.recv_rtp_sink_0 rtpbin. !
> rtpL16depay ! audioconvert ! directsoundsink udpsrc
> uri=udp://224.0.1.5:5003 ! rtpbin.recv_rtcp_sink_0
> rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=224.0.1.5 sync=false
> async=false
>
> I think it must have something to do with the 'caps' or a buffer
> problem. Any ideas?
>
> Thanks!
>
> Best,
>
> mat
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>



-- 
www.mediamagictechnologies.com (Gstreamer, ffmpeg, Red5, Streaming)
twitter handle: @gst_kaps
http://www.linkedin.com/in/kapilagrawal
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