共计 13079 个字符，预计需要花费 33 分钟才能阅读完成。
一、简介
WebRTC 概念
WebRTC 是由 Google 主导的，由一组规范、协定和 JavaScript API 组成，用于实现浏览器之间（端到端之间）的音频、视频及数据共享。WebRTC 不须要装置任何插件，通过简略的 JavaScript API 就能够使得实时通信变成一种规范性能。
为什么应用 webrtc
当初各大浏览器以及终曾经逐步加大对 WebRTC 技术的反对。下图是 webrtc 官网给出的当初曾经提供反对了的浏览器和平台。
二、srs 启动
webrtc 须要 srs 的最低版本：SRS4.0.14，为了拉取到最新版本的 srs，咱们首先做的就是获取到最新的 srs 代码，装置 git 步骤如下：
centos 装置 git
# 装置依赖
yum install curl-devel expat-devel gettext-devel openssl-devel zlib-devel
yum install gcc-c++ perl-ExtUtils-MakeMaker
#查看 yum 源仓库 git 信息
yum info git
#移除默认装置 git
yum remove git
#装置 git
yum install git
#查看 git 版本
git --versionwebrtc 须要 srs 的最低版本：SRS4.0.14，所以咱们部署的 srs 的时候版本须要大于 SRS4.0.14，如果应用 git 获取分支能够应用如下命令(进入到 srs 的 git 目录）：
# 查看以后 git 分支信息(默认分支 * 指定, 以后为 3.0)
git branch -v
#rtc 在 4.0 或 develop 分支上能够拉取到, 切换到 4.0
git checkout 4.0release
#再次查看以后所处分支
git branch -v
#如果要查看所有公布的 git 版本能够应用
git tagwebrtc 是默认反对的（–rtc=on），所以间接编译即可：
./configure --with-hls --with-ssl --with-http-server --with-http-callback --with-http-api --with-ingest --with-stream-caster && make 而后，咱们能够应用默认的 rtc 配置 (conf/rtc.conf) 跑起来：
cd srs-4.0.39/trunk
./objs -c conf/rtc.conf默认 rtc.conf 配置如下
listen 1935;
max_connections 1000;
srs_log_tank console;
srs_log_file ./objs/srs.log;
daemon off;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
#rtc 用到的 api 服务端口
http_api {
enabled on;
listen 1985;
}
stats {network 0;}
#rtc 的配置
rtc_server {
enabled on;
# Listen at udp://8000
listen 8000;
#
# The $CANDIDATE means fetch from env, if not configed, use * as default.
#
# The * means retrieving server IP automatically, from all network interfaces,
# @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
#拉取流地址: 应用本机地址或如下配置
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
#vhost 关上启用 rtc
rtc {
enabled on;
bframe discard;
}
}启动后，能够看到 rtc 监听的端口信息
三、推送 rtmp 视频
启动后咱们须要采集本地的音视频，而后推送至 srs 中，srs 通过协定转换生成 webrtc 协定
咱们推送 rtmp 的流到 srs 中
rtmp://192.168.12.187:1935/live/1我这里应用 obs，你也能够应用 ffmpeg 推送 rtmp
ffmpeg -f dshow -i video="HD Camera":audio="麦克风阵列 (Realtek(R) Audio)" -vcodec libx264 -x264opts "bframes=0" -r 25 -g 25 -preset:v ultrafast -tune:v zerolatency -codec:a aac -ac 2 -ar 44100 -f flv rtmp://192.168.50.150:1935/live/1 推送流胜利之后，咱们能够应用 srs 自带的 rtc_player 播放器进行播放，间接申请 srs 服务的 8080 端口即可
http://192.168.12.187:8080/players/rtc_player.html
四、webrtc 播放
能够看到咱们应用 webrtc 协定就能够播放该视频流了，视频流地址：webrtc://srs
webrtc://192.168.12.187/live/1chrome 的 rtc 调式模式关上(如调试黑屏问题 candidate)：
chrome://webrtc-internals
webrtc 调试申请参数（能够看到是通过 api 接口的 1985 端口收回的，正式咱们启用的）
webrtc 提供的播放接口为：
http://192.168.12.187:1985/rtc/v1/play/
http://192.168.12.187:1985/rtc/v1/play/申请参数：
{
"api": "http://192.168.12.187:1985/rtc/v1/play/",
"streamurl": "webrtc://192.168.12.187/live/1",
"clientip": null,
"sdp": "v=0\r\no=- 8774286378837017703 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:BLet\r\na=ice-pwd:cbGk3IU+Qdmq87i4JEEp1KGk\r\na=ice-options:trickle\r\na=fingerprint:sha-256 22:2E:04:F1:E1:12:0F:35:7B:16:C2:51:FC:79:D6:B8:49:78:6E:FE:B5:1A:8E:0F:91:85:61:C4:50:47:14:18\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 122 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:BLet\r\na=ice-pwd:cbGk3IU+Qdmq87i4JEEp1KGk\r\na=ice-options:trickle\r\na=fingerprint:sha-256 22:2E:04:F1:E1:12:0F:35:7B:16:C2:51:FC:79:D6:B8:49:78:6E:FE:B5:1A:8E:0F:91:85:61:C4:50:47:14:18\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 profile-id=0\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 VP9/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 profile-id=2\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:122 VP9/90000\r\na=rtcp-fb:122 goog-remb\r\na=rtcp-fb:122 transport-cc\r\na=rtcp-fb:122 ccm fir\r\na=rtcp-fb:122 nack\r\na=rtcp-fb:122 nack pli\r\na=fmtp:122 profile-id=1\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=fmtp:121 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcp-fb:124 transport-cc\r\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=124\r\na=rtpmap:123 H264/90000\r\na=rtcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\r\na=rtpmap:118 rtx/90000\r\na=fmtp:118 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\n"
}webrtc 播放申请回复：
{
"code": 0,
"server": 377766,
"sdp": "v=0\r\no=SRS/4.0.39(Leo) 30292192 2 IN IP4 0.0.0.0\r\ns=SRSPlaySession\r\nt=0 0\r\na=ice-lite\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS live/1\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:75m4425y\r\na=ice-pwd:5k49gz6h45c56i27f1ltb735z9667dad\r\na=fingerprint:sha-256 37:69:01:ED:46:E4:E4:CC:1F:8A:71:50:2D:27:57:C8:7E:E3:0E:F6:12:3D:F2:84:F9:28:56:39:3F:81:6E:13\r\na=setup:passive\r\na=mid:0\r\na=sendonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:111 opus/48000/2\r\na=ssrc:1271938135 cname:1wty44643861r274\r\na=ssrc:1271938135 label:audio-mt940p61\r\na=candidate:0 1 udp 2130706431 192.168.12.187 8000 typ host generation 0\r\na=candidate:1 1 udp 2130706431 172.17.0.1 8000 typ host generation 0\r\na=candidate:2 1 udp 2130706431 10.224.98.128 8000 typ host generation 0\r\nm=video 9 UDP/TLS/RTP/SAVPF 102\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:75m4425y\r\na=ice-pwd:5k49gz6h45c56i27f1ltb735z9667dad\r\na=fingerprint:sha-256 37:69:01:ED:46:E4:E4:CC:1F:8A:71:50:2D:27:57:C8:7E:E3:0E:F6:12:3D:F2:84:F9:28:56:39:3F:81:6E:13\r\na=setup:passive\r\na=mid:1\r\na=sendonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:102 H264/90000\r\na=ssrc:1271938136 cname:1wty44643861r274\r\na=ssrc:1271938136 label:video-lz539zd4\r\na=candidate:0 1 udp 2130706431 192.168.12.187 8000 typ host generation 0\r\na=candidate:1 1 udp 2130706431 172.17.0.1 8000 typ host generation 0\r\na=candidate:2 1 udp 2130706431 10.224.98.128 8000 typ host generation 0\r\n",
"sessionid": "75m4425y:BLet"
}通过测试 webrtc 在动画场景下是呈现花屏的，这个应该不是 udp 导致的（我去掉 bframe discard 后不花屏，然而呈现杂音和卡顿），具体的 webrtc 配置咱们能够去 srs 的 4.0 版本上查看 full.conf 看具体阐明：
https://github.com/ossrs/srs/blob/4.0release/trunk/conf/full.conf我截取了 webrtc 局部的配置和阐明，详见如下：
#############################################################################################
# WebRTC server section
#############################################################################################
rtc_server {
# Whether enable WebRTC server.
# default: off
enabled on;
# The udp listen port, we will reuse it for connections.
# default: 8000
listen 8000;
# The exposed candidate IPs, response in SDP candidate line. It can be:
# * Retrieve server IP automatically, from all network interfaces.
# eth0 Retrieve server IP by specified network interface name. # TODO: Implements it.
# $CANDIDATE Read the IP from ENV variable, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
# x.x.x.x A specified IP address or DNS name, which can be access by client such as Chrome.
# You can specific more than one interface name:
# eth0 eth1 Use network interface eth0 and eth1. # TODO: Implements it.
# Also by IP or DNS names:
# 192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
# And by multiple ENV variables:
# $CANDIDATE $EIP # TODO: Implements it.
# default: *
candidate *;
# The IP family filter for candidate, it can be:
# ipv4 Filter IP v4 candidates.
# ipv6 Filter IP v6 candidates.
# all Filter all IP v4 or v6 candidates.
# For example, if set to ipv4, we only use the IPv4 address as candidate.
# default: ipv4
ip_family ipv4;
# Whether use ECDSA certificate.
# If not, use RSA certificate.
# default: on
ecdsa on;
# Whether encrypt RTP packet by SRTP.
# @remark Should always turn it on, or Chrome will fail.
# default: on
encrypt on;
# We listen multiple times at the same port, by REUSEPORT, to increase the UDP queue.
# Note that you can set to 1 and increase the system UDP buffer size by net.core.rmem_max
# and net.core.rmem_default or just increase this to get larger UDP recv and send buffer.
# default: 1
reuseport 1;
# Whether merge multiple NALUs into one.
# @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
# default: off
merge_nalus off;
# Whether enable the perf stat at http://localhost:1985/api/v1/perf
# default: on
perf_stat on;
# The queue length, in number of mmsghdr, in messages.
# For example, 30 means we will cache 30K messages at most.
# If exceed, we will drop messages.
# @remark Each reuseport use a dedicated queue, if queue is 2000, reuseport is 4,
# then system queue is 2000*4 = 8k, user can incrase reuseport to incrase the queue.
# default: 2000
queue_length 2000;
# The black-hole to copy packet to, for debugging.
# For example, when debugging Chrome publish stream, the received packets are encrypted cipher,
# we can set the publisher black-hole, SRS will copy the plaintext packets to black-hole, and
# we are able to capture the plaintext packets by wireshark.
black_hole {
# Whether enable the black-hole.
# default: off
enabled off;
# The black-hole address for session.
addr 127.0.0.1:10000;
}
}vhost 下 rtc 启用配置：
vhost rtc.vhost.srs.com {
rtc {
# Whether enable WebRTC server.
# default: off
enabled on;
# The strategy for bframe.
# keep Keep bframe, which may make browser with playing problems.
# discard Discard bframe, maybe cause browser with little problems.
# default: keep
bframe discard;
# The strategy for aac audio.
# transcode Transcode aac to opus.
# discard Discard aac audio packet.
# default: transcode
aac transcode;
# The timeout in seconds for session timeout.
# Client will send ping(STUN binding request) to server, we use it as heartbeat.
# default: 30
stun_timeout 30;
# The strick check when process stun.
# default: off
stun_strict_check on;
# The role of dtls when peer is actpass: passive or active
# default: passive
dtls_role passive;
# The version of dtls, support dtls1.0, dtls1.2, and auto
# default: auto
dtls_version auto;
# Drop the packet with the pt(payload type), 0 never drop.
# default: 0
drop_for_pt 0;
}
# whether enable min delay mode for vhost.
# default: on, for RTC.
min_latency on;
play {# set the MW(merged-write) latency in ms.
# @remark For WebRTC, we enable pass-timestamp mode, so we ignore this config.
# default: 0 (For WebRTC)
mw_latency 0;
# Set the MW(merged-write) min messages.
# default: 0 (For Real-Time, min_latency on)
# default: 1 (For WebRTC, min_latency off)
mw_msgs 0;
}
# For NACK.
nack {
# Whether support NACK.
# default: on
enabled on;
}
# For TWCC.
twcc {
# Whether support TWCC.
# default: on
enabled on;
}
}也能够在客户端指定 webrtc 服务器 wlan 的 ip(如果 docker 获取不到 ip 能够通过设置环境变量形式 $CANDIDATE)：
webrtc://192.168.12.187/live/1?eip=192.168.12.187如果应用 docker 测试能够应用拉取对应镜像
# 查看公布的所有的版本
git tag
#通过查看到的版本拉取对应的 docker 镜像(也能够应用杭州的镜像)
docker pull osrs/srs:v4.0.34查看源码和对应文档能够进入 srs-docker 版本，切换到 rts 分支：
https://github.com/ossrs/srs-docker/tree/rtc源码获取、单干、技术交换请获取如下联系方式：
QQ 交换群：961179337
微信账号：lixiang6153
公众号：IT 技术快餐
电子邮箱：lixx2048@163.com
正文完
