Good morning, I'm trying to live transcode input (mp2/mpeg2video) to libamr_nb/ffh263(Supported formats in mobile devices), the problem is that when launching ffmpeg:
ffmpeg -i rtp://0.0.0.0?localport=2020 -acodec libamr_nb -ar 8000 -ac 1 -ab 12.2k -f rtp rtp://192.168.50.147:8082 -an -vcodec h263p -flags aic -r 14 -b 20k -s 176*144 -f rtp rtp://192.168.50.147:8080 I have this very poor SDP: v=0 o=- 0 0 IN IPV4 127.0.0.1 t=0 0 s=No Name a=tool:libavformat m=audio 8082 RTP/AVP 96 c=IN IP4 192.168.50.42 m=video 8080 RTP/AVP 96 c=IN IP4 192.168.50.42 and since 96 is a dynamic type, I have to add specifications to the streaming server to make it able to deliver successful RTSP session (using: a=rtpmap:96 H263-2000/90000 and a=rtpmap:96 AMR/8000/1). I attached the SDP file I'm using. Realplayer is buffering until 100% but not displaying video, and the audio is very fast as if the sampling frequency is incorrect. Help please! -- Mahmoud M'HIRI [EMAIL PROTECTED] Phone: Under request
ch.sdp
Description: application/sdp
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