2018-03-13 8:15 GMT+01:00, lagavulin2016 <lagavulin2...@naver.com>: > Hello. I'm trying to convert voip pcap to wav. > voip pcap has a few bidirectional call and includes sip and rtp packets. > (for example, https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP) > So I created libavformat/voip_pcap.c which is similar to libavformat/rtsp.c > with pcap file parsing instead of networking. > It seems to work except g729 codec. > 1. g729 codec is not recognized because in rtp_payload_types from > libavformat/rtp.c > "G729" codec_id is AV_CODEC_ID_NONE. If I change to AV_CODEC_ID_G729, it > is recognized well. > Is this intentionally none? or g729 in rtp is not supported?
I believe a patch to support G.729 over rtp would be very welcome. Do you know how such a patch could be tested? > 2. g729 annex b has 2-bytes Silence Insertion Descriptor(SID) frame > but ffmpeg doesn't seem to support this. Can you provide a real-life sample of G.729B? Carl Eugen _______________________________________________ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user
