I am trying to get the audio and video from a WebRTC stream and handle it (transcode or dump) with ffmpeg on ubuntu server. I have naively expected it to simply interpret the sdp offered by WebRTC, but was mistaken. I suspect ffmpeg is not capable of signaling back the answer sdp and it is must be done manually. Here is an offer sdp:
v=0
o=Mozilla-SIPUA-33.1 3662 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:5e0a74d1
a=ice-pwd:7446c0eb445117d0018ca2afc5f3ca54
a=fingerprint:sha-256 76:1B:19:CE:3F:41:C5:4D:64:E6:FD:72:D2:FC:42:E1:98:D4:0F:2B:73:AE:C7:F4:2C:73:2C:E7:9E:DC:6A:EB
m=audio 23063 RTP/SAVPF 109 0 8 101
c=IN IP4 188.253.146.208
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 192.168.1.100 4130 typ host
a=candidate:1 1 UDP 1692467199 188.253.146.208 23063 typ srflx raddr 192.168.1.100 rport 4130
a=candidate:0 2 UDP 2128609534 192.168.1.100 4131 typ host
a=candidate:1 2 UDP 1692467198 188.253.146.208 23064 typ srflx raddr 192.168.1.100 rport 4131
a=rtcp-mux
m=video 23065 RTP/SAVPF 120 126 97
c=IN IP4 188.253.146.208
a=rtpmap:120 VP8/90000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01f
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=setup:actpass
a=candidate:0 1 UDP 2128609535 192.168.1.100 4132 typ host
a=candidate:1 1 UDP 1692467199 188.253.146.208 23065 typ srflx raddr 192.168.1.100 rport 4132
a=candidate:0 2 UDP 2128609534 192.168.1.100 4133 typ host
a=candidate:1 2 UDP 1692467198 188.253.146.208 23066 typ srflx raddr 192.168.1.100 rport 4133
a=rtcp-mux
Is anybody have succeeded in it? Thanks.
