Hi, Stefan! Thanks for you reply. So here is the pipe with two streams that dont work together: gst-launch -v udpsrc port=5000 caps="application/x-rtp,media=audio,payload=3,clock-rate=8000,channels=1,encoding-name=GSM" ! .recv_rtp_sink_0 gstrtpbin ! rtpgsmdepay ! gsmdec ! audioconvert ! pulsesink gstrtpbin name=rtpbin autoaudiosrc ! gsmenc ! rtpgsmpay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! identity ! udpsink port=5000 host=127.0.0.1 I do not get any error when running that pipe! But I cant hear anything when running it. However, when I separate the streams, like below, and run then on different terminals, that works(but that is like two pipes and I dont want two pipes). The audio is captured and sent and I hear myself as I talk. receiver stream: gst-launch -v udpsrc port=5000 caps="application/x-rtp,media=audio,payload=3,clock-rate=8000,channels=1,encoding-name=GSM" ! .recv_rtp_sink_0 gstrtpbin ! rtpgsmdepay ! gsmdec ! audioconvert ! pulsesink sender stream: gst-launch -v gstrtpbin name=rtpbin autoaudiosrc ! gsmenc ! rtpgsmpay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! identity ! udpsink port=5000 host=127.0.0.1 As I said before, I do not get any error when running those streams in one pipeline, but I cant hear any audio sent through udp... <div class="gmail_quote">2009/10/30 Rajesh Marathe <<a href="mailto:rmarathe@i-rode.com" target="_blank">rmarathe@i-rode.com</a>> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Hi Stefan, Sorry to diverge slightly from the topic, but, I do have a query on two streams. I tried doing this using gstreamer: - Create two different pipelines for two streams. (Create different elements for each pipeline but these elements refer to same hardware units). - Play stream1 while stream2 is in PAUSED state. - After few seconds, PAUSE stream1 and move stream2 to PLAY state. - I notice that stream1 plays and pauses but stream2 does not play actually. Will the above scenario work and if yes, do you see anything that needs done apart from the above ? thanks and regards, Rajesh Marathe. <div><div></div><div> On 10/30/2009 12:58 AM, Stefan Kost wrote: > Marcelo de Sá Mendoza schrieb: > >> Hi >> >> Is possible to have two streams( 1 sender 1 receiver ) or more, of audio >> in one pipeline only? I'm trying to do that and I getting frustrated. I >> made two pipelines for each stream and worked cool, I hear the audio >> received from the host and the host hear what I talk. The thing is that >> I need to do that in ONE pipeline, but I can't. I know that two streams >> are possible in one pipeline, cause when I do the test below, it works: >> >> gst-launch -v filesrc location=notice.html ! identity ! udpsink >> host=127.0.0.1 port=6000 udpsrc port=6000 ! identity ! filesink >> location=notice2.html >> >> but when I change to pulsesrc/pulsesink that doesnt work. Works only in >> diferent pipelines. There's something else that I need to do? >> > You can have as many streams in a pipe as you want. The above example is quite > uncomplete though. You probably want to use rtp for sending audio over the > network. Also if something does not work, tell us what errors you get. Otherwise > we can't help. > > Stefan > > >> Thanks for the atention. >> >> >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------------ >> Come build with us! 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The BlackBerry(R) Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9 - 12, 2009. Register now! > <a href="http://p.sf.net/sfu/devconference" target="_blank">http://p.sf.net/sfu/devconference</a> > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> > <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> > > ------------------------------------------------------------------------------ Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! <a href="http://p.sf.net/sfu/devconference" target="_blank">http://p.sf.net/sfu/devconference</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> </div></div></blockquote></div>