webrtc的sdp分offer和answer两个，以offer为例，简单分析 ，其实就2个m（media）就可以了.
"sdp" : "
v=0 //version 版本
o=- 6547339724864950015 2 IN IP4 127.0.0.1 //origin 源 IN为internet
s=- //session 会话
t=0 0 //time 活动时间
a=group:BUNDLE audio video //attribute 属性
a=msid-semantic: WMS stream_label //以上为总体配置
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126 //media 音频配置，优先111 payload
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:qZdD
a=ice-pwd:n+USATgn6aNU9gwAgYUAKhTZ
a=ice-options:trickle
a=fingerprint:sha-256 30:56:8B:C6:82:6A:65:21:28:6F:B3:4D:4D:C9:32:A1:D1:C5:57:41:D0:6B:47:E9:71:4B:21:AA:74:81:DE:7C
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux //rtp和rtcp在一个端口发送
a=rtpmap:111 opus/48000/2 //优先为opus
a=rtcp-fb:111 transport-cc //拥塞控制
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:4030990412 cname:8qkjGJ0/dox9q4jP
a=ssrc:4030990412 msid:stream_label audio_label
a=ssrc:4030990412 mslabel:stream_label
a=ssrc:4030990412 label:audio_label
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127 123 125 122 124 //开始视频配置，96为优先协议
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:qZdD
a=ice-pwd:n+USATgn6aNU9gwAgYUAKhTZ
a=ice-options:trickle
a=fingerprint:sha-256 30:56:8B:C6:82:6A:65:21:28:6F:B3:4D:4D:C9:32:A1:D1:C5:57:41:D0:6B:47:E9:71:4B:21:AA:74:81:DE:7C
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=sendrecv
a=rtcp-mux //rtp和rtcp在一个端口发送
a=rtcp-rsize //rtcp缩小 rfc 5506
a=rtpmap:96 VP8/90000 //vp8为优先协议
a=rtcp-fb:96 goog-remb //带宽估计反馈，动态调整码率用的
a=rtcp-fb:96 transport-cc //拥塞控制
a=rtcp-fb:96 ccm fir //关键帧重传
a=rtcp-fb:96 nack //丢包重传
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 H264/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:123 rtx/(stunport.cc:387): Jingle:Port[0x7f3b94027f10:audio:1:0:local:Net[eth0:10.25.16.0/24:Unknown]]: Starting STUN host lookup for stun.l.google.com:19302
90000
a=fmtp:123 apt=127
a=rtpmap:125 red/90000
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=125
a=rtpmap:124 ulpfec/90000
a=ssrc-group:FID 301401763 1154011058 //以ssrc区分rtp rtcp video和audio
a=ssrc:301401763 cname:8qkjGJ0/dox9q4jP
a=ssrc:301401763 msid:stream_label video_label
a=ssrc:301401763 mslabel:stream_label
a=ssrc:301401763 label:video_label
a=ssrc:1154011058 cname:8qkjGJ0/dox9q4jP
a=ssrc:1154011058 msid:stream_label video_label
a=ssrc:1154011058 mslabel:stream_label
a=ssrc:1154011058 label:video_label",
"type" : "offer"