> > 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.
_______________________________________________ Libav-user mailing list Libavfirstname.lastname@example.org http://ffmpeg.org/mailman/listinfo/libav-user