<html> <head> <style> .hmmessage P { margin:0px; padding:0px } body.hmmessage { font-size: 10pt; font-family:Verdana } </style> </head> <body class='hmmessage'> The command I put in the last e-mail was wrong. The correct one is the following. I am thinking that it may be a reference for others. That is why i wanted to send the correct one.<br><br>gst-launch-0.10 -v udpsrc port=5002 caps="application/x-rtp,media=(string)audio,payload=(int)96, rate=(int)8000, encoding-name=(string)PCMA" ! queue ! rtppcmadepay ! 'audio/x-alaw, rate=(int)8000, channels=(int)1' ! queue ! avimux ! filesink location=audio.avi sync=false<br><br>Thank you again.<br><br>- Zelalem S. <br><br><hr id="stopSpelling">From: zelalems@hotmail.com<br>To: gstreamer-devel@lists.sourceforge.net<br>Date: Thu, 11 Jun 2009 13:03:49 +0300<br>Subject: Re: [gst-devel] Problem with Recording Audio from a Network<br><br> <style> .ExternalClass .EC_hmmessage P {padding:0px;} .ExternalClass body.EC_hmmessage {font-size:10pt;font-family:Verdana;} </style> Hi Roope and Jyoti, thank you very much for your advice. It is now working. I copied the caps from the output of the sender for the caps (that was a very good advice). Anyway, I used the x-rtp for the udp caps and x-alaw in between the depayloader and the avimuxer. The following is the command:<br><br>gst-launch-0.10 -v udpsrc port=5002 caps="application/x-rtp,media=(string)audio,payload=(int)96, rate=(int)8000, encoding-name=(string)PCMA" ! queue ! rtppcmadepay ! 'audio/x-alaw, rate=(int)8000, channels=(int)1' ! queue ! alawdec ! audioconvert ! alsasink<br><br>I hope, i will not face any problem with mixing the video and audio. <br><br>Thank you again.<br><br>- Zelalem S. <br><br><br><hr id="EC_stopSpelling">From: roope.jarvinen@nokia.com<br>To: gstreamer-devel@lists.sourceforge.net<br>Date: Thu, 11 Jun 2009 10:15:37 +0200<br>Subject: Re: [gst-devel] Problem with Recording Audio from a Network<br><br> <style> .ExternalClass .EC_hmmessage P {padding-right:0px;padding-left:0px;padding-bottom:0px;padding-top:0px;} .ExternalClass BODY.EC_hmmessage {font-size:10pt;font-family:Verdana;} </style> <div dir="ltr" align="left"><font color="#0000ff" face="Arial"></font> </div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial">Hi Zelalem,</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial"></font></span> </div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial">You can use rtppcmapay/depay and rtppcmupay/depay elements with alaw and ulaw.</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial"></font></span> </div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial">Correct type for alaw is audio/x-alaw and not audio/x-alaw-int.</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial"></font></span> </div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial">--Roope</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_800241108-11062009"><font color="#0000ff" face="Arial"></font></span> </div><br> <blockquote style="padding-left: 5px; margin-left: 5px; margin-right: 0px;"> <div class="EC_EC_OutlookMessageHeader" dir="ltr" align="left" lang="en-us"> <hr> <font face="Tahoma"><b>From:</b> ext Zelalem Sintayehu [mailto:zelalems@hotmail.com] <br><b>Sent:</b> 11 June, 2009 11:08<br><b>To:</b> gstreamer-devel@lists.sourceforge.net<br><b>Subject:</b> Re: [gst-devel] Problem with Recording Audio from a Network<br></font><br></div> <div></div>Hi Roope, you are right, it allows alaw,mulaw,ac3,mpeg and raw. But alaw and mulaw don't have payloader and depayloader. Anyway, I tried to recieve without depayloader but it still produced the same error. The following is what  tried. Please help me.<br><br>gst-launch-0.10 -v udpsrc port=5002 caps="audio/x-rtp,rate=1000,channels=1,depth=8" ! queue ! audio/x-alaw-int,rate=1000,channels=1,depth=8 ! avimux ! filesink location=audio.avi sync=false .   I also changed the caps for udpsrc with "audio/x-alaw-int,rate=1000,channels=1,depth=8" but didn't work.<br><br>Thank you.<br><br>- Zelalem S. <br><br> <hr id="EC_EC_stopSpelling"> From: roope.jarvinen@nokia.com<br>To: gstreamer-devel@lists.sourceforge.net<br>Date: Thu, 11 Jun 2009 09:35:58 +0200<br>Subject: Re: [gst-devel] Problem with Recording Audio from a Network<br><br> <style> .ExternalClass .EC_hmmessage P {padding-right:0px;padding-left:0px;padding-bottom:0px;padding-top:0px;} .ExternalClass BODY.EC_hmmessage {font-size:10pt;font-family:Verdana;} </style> <div dir="ltr" align="left"><font color="#0000ff" face="Arial"></font> </div> <div dir="ltr" align="left"><font color="#0000ff" face="Arial"></font> </div> <div dir="ltr" align="left"><span class="EC_EC_EC_405453407-11062009"><font color="#0000ff" face="Arial">Hi,</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_EC_405453407-11062009"><font color="#0000ff" face="Arial"></font></span> </div> <div dir="ltr" align="left"><span class="EC_EC_EC_405453407-11062009"><font color="#0000ff" face="Arial">You cannot mux gsm-encoded audio into AVI container. Check avimux description for allowed formats.</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_EC_405453407-11062009"><font color="#0000ff" face="Arial"></font></span> </div> <div dir="ltr" align="left"><span class="EC_EC_EC_405453407-11062009"><font color="#0000ff" face="Arial">--Roope</font></span></div> <div dir="ltr" align="left"><span class="EC_EC_EC_405453407-11062009"></span> </div><br> <blockquote style="padding-left: 5px; margin-left: 5px; margin-right: 0px;"> <div class="EC_EC_EC_OutlookMessageHeader" dir="ltr" align="left" lang="en-us"> <hr> <font face="Tahoma"><b></b><br></font><br></div> <div></div>Hi Jyoti, thank you for your prompt response. I added the following caps statement, but it is still the same. The following is the modified receiver side code.<br><br>gst-launch-0.10 -v udpsrc port=5002 caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)GSM,encoding-params=(string)1,octet-align=(string)1" ! queue ! rtpgsmdepay ! audio/x-raw-int,rate=8000,channels=1,depth=8 ! avimux ! filesink location=audio.avi sync=false<br><br>The error is the same: "WARNING: erroneous pipeline: could not link rtpgsmdepay0 to avimux0"<br><br>Thank you.<br><br>- Zelalem S. <br>-----------------------------------<br>From: jyoti.d@allaboutif.com<br>To: gstreamer-devel@lists.sourceforge.net<br>Subject: Re: [gst-devel] Problem with Recording Audio from a Network<br><br>You should set caps property on udpsrc element at receiver side.<br><br></blockquote><a href="http://www.microsoft.com/windows/windowslive/"><br></a> </blockquote><br><hr>What can you do with the new Windows Live? <a href="http://www.microsoft.com/windows/windowslive/default.aspx">Find out</a><br /><hr />See all the ways you can stay connected <a href='http://www.microsoft.com/windows/windowslive/default.aspx' target='_new'>to friends and family</a></body> </html>