Re: [Libav-user] g729 in rtp
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: > 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 ___ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user
Re: [Libav-user] g729 in rtp
2018-03-14 6:20 GMT+01:00, lagavulin2016: > Previous post was somewhat html-formatted. I re-posted. It did not really improve: http://ffmpeg.org/pipermail/libav-user/2018-March/011003.html In addition, it is now also unreadable in html-aware mail clients... Carl Eugen ___ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user
Re: [Libav-user] g729 in rtp
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. ___ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user
Re: [Libav-user] g729 in rtp
> 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. ___ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user
Re: [Libav-user] g729 in rtp
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? > 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
[Libav-user] g729 in rtp
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.cwith 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? ___ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user