Check out the caps between your rtpgsmdepay and avimux. <br>Run server side with -v option and copy the caps on the gsmenc and apply those caps between rtpgsmdepay & avimux.<br><br><br><div class="gmail_quote">On Thu, Jun 11, 2009 at 12:53 PM, Zelalem Sintayehu <span dir="ltr"><<a href="mailto:zelalems@hotmail.com">zelalems@hotmail.com</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> <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: <a href="mailto:jyoti.d@allaboutif.com" target="_blank">jyoti.d@allaboutif.com</a><br> To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a><br>Subject: Re: [gst-devel] Problem with Recording Audio from a Network<div class="im"><br><br>You should set caps property on udpsrc element at receiver side.<br> <br></div><div><div class="im">On Wed, Jun 10, 2009 at 7:26 PM, Zelalem Sintayehu <span dir="ltr"><<a href="mailto:zelalems@hotmail.com" target="_blank">zelalems@hotmail.com</a>></span> wrote:<br> </div><blockquote style="padding-left: 1ex;"> <div><div class="im"> Hi all, I was trying to record audio and video over the network. I wanted to check that separately and succeeded with the video recording, but i couldn't record the audio part. BTW, I want to record into avi file and I used h263p for the video and wanted to use either amr or gsm for the audio. I hope this is possible. The following is my code both from teh sender and reciever side.<br> <br>Sender:<br>gst-launch-0.10 -v alsasrc ! queue ! audio/x-raw-int,rate=8000,channels=1,depth=8 ! audioconvert ! gsmenc ! rtpgsmpay ! queue ! udpsink port=5002<br><br>Receiver:<br><br>gst-launch-0.10 -v udpsrc port=5002 ! queue ! rtpgsmdepay ! audio/x-raw-int,rate=8000,channels=1,depth=8 ! avimux ! filesink location=audio.avi sync=false<br> <br>and I got the following error and it terminates. "WARNING: erroneous pipeline: could not link rtpgsmdepay0 to avimux0. By the way, when i store both the video later, should i still put "sync=false" (the last statement in the reciever code)? <br> <br><br>Please help me.<br><br></div><hr></div></blockquote></div><div class="im"><br><br><hr>Windows Live™: Keep your life in sync. <a href="http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009" target="_blank">Check it out!</a></div> </div> <br>------------------------------------------------------------------------------<br> Crystal Reports - New Free Runtime and 30 Day Trial<br> Check out the new simplified licensing option that enables unlimited<br> royalty-free distribution of the report engine for externally facing<br> server and web deployment.<br> <a href="http://p.sf.net/sfu/businessobjects" target="_blank">http://p.sf.net/sfu/businessobjects</a><br>_______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a><br> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br> <br></blockquote></div><br>