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?
2. g729 annex b has 2-bytes Silence Insertion Descriptor(SID) frame
but ffmpeg doesn't seem to support this.
Is there any plan to support g729(and annex b)? or am I doing something wrong?
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