[Libav-user] g729 in rtp

유현우 lagavulin2016 at naver.com
Thu Mar 15 01:55:08 EET 2018

I'm replying with another mail client now. sorry for duplicate.
1. 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. g729a pcap sample is in the wireshark sample page I posted, but g729b is hard to find on the internet :(
  I'll search more.
2018-03-13 8:15 GMT+01:00, lagavulin2016 <lagavulin2016 at 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 at ffmpeg.org
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