Previous post was somewhat html-formatted. I re-posted. > 2018-03-13 8:15 GMT+01:00, lagavulin2016 : > > 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? Well..my demuxer is not ready to commit yet... but simple question is.. AV_CODEC_ID_NONE means g729 over rtp is not supported? or codec is recognized elsewhere? Because g729 is very common in voip, so I couldn't believe ffmpeg doesn't support g729 over rtp. If it is really not supported, I would start thinking contribute. > > > 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 g729a pcap sample is in the wireshark sample page I posted, but g729b is hard to find on the internet :( I'll search more.

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