版本：6.0 (Stable) ✅

RTMP

RTMP是直播的事实标准，这么多年以来一直是使用最广泛的直播协议。

然而Adobe公司没有一直更新RTMP协议，也没有提交给标准组织比如RFC，因此很多新功能都没有支持，比如HEVC或Opus。直到2023.03，终于Enhanced RTMP项目建立，开始支持了HEVC和AV1， SRS和OBS已经支持了基于Enhanced RTMP的HEVC编码。

在流的制作方面，最近几年，SRT、WebRTC和SRT增长迅速，很多设备都支持了SRT和RIST协议。你也可以用WebRTC做直播。

在流的分发上，HLS是使用最广泛的协议，所有CDN和设备都支持，比如PC，iOS，Android或平板电脑。当然HLS延迟比较大(3~5s+)，你可以选择HTTP-FLV，HTTP-TS或者WebRTC，如果需要降低延迟。

至今为止，在内容制作领域，RTMP还是使用最广泛的协议。比如你可以用OBS推流到B站、视频号或快手。如果要对接一个广播设备，或者推流到某个平台，那么RTMP是最好的选择，几乎都会支持。

Usage

SRS内置RTMP的支持，可以用docker或者从源码编译:

docker run --rm -it -p 1935:1935 registry.cn-hangzhou.aliyuncs.com/ossrs/srs:5 \ ./objs/srs -c conf/rtmp.conf

使用 FFmpeg(点击下载) 或 OBS(点击下载) 推流：

ffmpeg -re -i ./doc/source.flv -c copy -f flv rtmp://localhost/live/livestream

打开下面的页面播放流（若SRS不在本机，请将localhost更换成服务器IP）:

RTMP (by VLC): rtmp://localhost/live/livestream

SRS支持将RTMP转换成其他协议，下面会详细描述。

Config

RTMP协议相关配置如下：

vhost __defaultVhost__ { # whether enable min delay mode for vhost. # for min latency mode: # 1. disable the publish.mr for vhost. # 2. use timeout for cond wait for consumer queue. # @see https://github.com/ossrs/srs/issues/257 # default: off (for RTMP/HTTP-FLV) # default: on (for WebRTC) min_latency off; # whether enable the TCP_NODELAY # if on, set the nodelay of fd by setsockopt # Overwrite by env SRS_VHOST_TCP_NODELAY for all vhosts. # default: off tcp_nodelay off; # the default chunk size is 128, max is 65536, # some client does not support chunk size change, # vhost chunk size will override the global value. # Overwrite by env SRS_VHOST_CHUNK_SIZE for all vhosts. # default: global chunk size. chunk_size 128; # The input ack size, 0 to not set. # Generally, it's set by the message from peer, # but for some peer(encoder), it never send message but use a different ack size. # We can chnage the default ack size in server-side, to send acknowledge message, # or the encoder maybe blocked after publishing for some time. # Overwrite by env SRS_VHOST_IN_ACK_SIZE for all vhosts. # Default: 0 in_ack_size 0; # The output ack size, 0 to not set. # This is used to notify the peer(player) to send acknowledge to server. # Overwrite by env SRS_VHOST_OUT_ACK_SIZE for all vhosts. # Default: 2500000 out_ack_size 2500000; # the config for FMLE/Flash publisher, which push RTMP to SRS. publish { # about MR, read https://github.com/ossrs/srs/issues/241 # when enabled the mr, SRS will read as large as possible. # Overwrite by env SRS_VHOST_PUBLISH_MR for all vhosts. # default: off mr off; # the latency in ms for MR(merged-read), # the performance+ when latency+, and memory+, # memory(buffer) = latency * kbps / 8 # for example, latency=500ms, kbps=3000kbps, each publish connection will consume # memory = 500 * 3000 / 8 = 187500B = 183KB # when there are 2500 publisher, the total memory of SRS at least: # 183KB * 2500 = 446MB # the recommended value is [300, 2000] # Overwrite by env SRS_VHOST_PUBLISH_MR_LATENCY for all vhosts. # default: 350 mr_latency 350; # the 1st packet timeout in ms for encoder. # Overwrite by env SRS_VHOST_PUBLISH_FIRSTPKT_TIMEOUT for all vhosts. # default: 20000 firstpkt_timeout 20000; # the normal packet timeout in ms for encoder. # Overwrite by env SRS_VHOST_PUBLISH_NORMAL_TIMEOUT for all vhosts. # default: 5000 normal_timeout 7000; # whether parse the sps when publish stream. # we can got the resolution of video for stat api. # but we may failed to cause publish failed. # @remark If disabled, HLS might never update the sps/pps, it depends on this. # Overwrite by env SRS_VHOST_PUBLISH_PARSE_SPS for all vhosts. # default: on parse_sps on; # When parsing SPS/PPS, whether try ANNEXB first. If not, try IBMF first, then ANNEXB. # Overwrite by env SRS_VHOST_PUBLISH_TRY_ANNEXB_FIRST for all vhosts. # default: on try_annexb_first on; # The timeout in seconds to disconnect publisher when idle, which means no players. # Note that 0 means no timeout or this feature is disabled. # Note that this feature conflicts with forward, because it disconnect the publisher stream. # Overwrite by env SRS_VHOST_PUBLISH_KICKOFF_FOR_IDLE for all vhosts. # default: 0 kickoff_for_idle 0; } # for play client, both RTMP and other stream clients, # for instance, the HTTP FLV stream clients. play { # whether cache the last gop. # if on, cache the last gop and dispatch to client, # to enabled fast startup for client, client play immediately. # if off, send the latest media data to client, # client need to wait for the next Iframe to decode and show the video. # set to off if requires min delay; # set to on if requires client fast startup. # Overwrite by env SRS_VHOST_PLAY_GOP_CACHE for all vhosts. # default: on gop_cache off; # Limit the max frames in gop cache. It might cause OOM if video stream has no IDR frame, so we limit to N # frames by default. Note that it's the size of gop cache, including videos, audios and other messages. # Overwrite by env SRS_VHOST_PLAY_GOP_CACHE_MAX_FRAMES for all vhosts. # default: 2500 gop_cache_max_frames 2500; # the max live queue length in seconds. # if the messages in the queue exceed the max length, # drop the old whole gop. # Overwrite by env SRS_VHOST_PLAY_QUEUE_LENGTH for all vhosts. # default: 30 queue_length 10; # about the stream monotonically increasing: # 1. video timestamp is monotonically increasing, # 2. audio timestamp is monotonically increasing, # 3. video and audio timestamp is interleaved/mixed monotonically increasing. # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format # however, some encoder cannot provides this feature, please set this to off to ignore time jitter. # the time jitter algorithm: # 1. full, to ensure stream start at zero, and ensure stream monotonically increasing. # 2. zero, only ensure stream start at zero, ignore timestamp jitter. # 3. off, disable the time jitter algorithm, like atc. # @remark for full, correct timestamp only when |delta| > 250ms. # @remark disabled when atc is on. # Overwrite by env SRS_VHOST_PLAY_TIME_JITTER for all vhosts. # default: full time_jitter full; # vhost for atc for hls/hds/rtmp backup. # generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0. # when atc is on, server delivery rtmp stream by absolute time. # atc is used, for instance, encoder will copy stream to master and slave server, # server use atc to delivery stream to edge/client, where stream time from master/slave server # is always the same, client/tools can slice RTMP stream to HLS according to the same time, # if the time not the same, the HLS stream cannot slice to support system backup. # # @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html # @see http://www.baidu.com/#wd=hds%20hls%20atc # # @remark when atc is on, auto off the time_jitter # Overwrite by env SRS_VHOST_PLAY_ATC for all vhosts. # default: off atc off; # whether use the interleaved/mixed algorithm to correct the timestamp. # if on, always ensure the timestamp of audio+video is interleaved/mixed monotonically increase. # if off, use time_jitter to correct the timestamp if required. # @remark to use mix_correct, atc should on(or time_jitter should off). # Overwrite by env SRS_VHOST_PLAY_MIX_CORRECT for all vhosts. # default: off mix_correct off; # whether enable the auto atc, # if enabled, detect the bravo_atc="true" in onMetaData packet, # set atc to on if matched. # always ignore the onMetaData if atc_auto is off. # Overwrite by env SRS_VHOST_PLAY_ATC_AUTO for all vhosts. # default: off atc_auto off; # set the MW(merged-write) latency in ms. # SRS always set mw on, so we just set the latency value. # the latency of stream >= mw_latency + mr_latency # the value recomment is [300, 1800] # @remark For WebRTC, we enable pass-by-timestamp mode, so we ignore this config. # default: 350 (For RTMP/HTTP-FLV) # Overwrite by env SRS_VHOST_PLAY_MW_LATENCY for all vhosts. # default: 0 (For WebRTC) mw_latency 350; # Set the MW(merged-write) min messages. # default: 0 (For Real-Time, min_latency on) # default: 1 (For WebRTC, min_latency off) # default: 8 (For RTMP/HTTP-FLV, min_latency off). # Overwrite by env SRS_VHOST_PLAY_MW_MSGS for all vhosts. mw_msgs 8; # the minimal packets send interval in ms, # used to control the ndiff of stream by srs_rtmp_dump, # for example, some device can only accept some stream which # delivery packets in constant interval(not cbr). # @remark 0 to disable the minimal interval. # @remark >0 to make the srs to send message one by one. # @remark user can get the right packets interval in ms by srs_rtmp_dump. # Overwrite by env SRS_VHOST_PLAY_SEND_MIN_INTERVAL for all vhosts. # default: 0 send_min_interval 10.0; # whether reduce the sequence header, # for some client which cannot got duplicated sequence header, # while the sequence header is not changed yet. # Overwrite by env SRS_VHOST_PLAY_REDUCE_SEQUENCE_HEADER for all vhosts. # default: off reduce_sequence_header on; } }

Note: 这里只是推流和拉流的配置，还有些其他的配置是在其他地方的，比如RTMP转HTTP-FLV或HTTP-TS等。

On Demand Live Streaming

有些场景下，是有需要播放时，才会邀请开始推流：

推流端连接到系统，但并不会推流到SRS。
播放器连接到系统，向系统请求播放流。
系统通知推流端，开始推流到SRS。
播放器从SRS拉流播放。

Note: 请注意系统是指你的业务系统，而不是SRS。

这就是我们所说的按需直播或按需推流。如果播放器停止拉流，会怎么样？

系统需要通知推流端停止推流。
或者，在最后一个播放器停止拉流时，SRS等待一定时间后断开推流。

推荐第2个解决方案，这样这个功能就非常容易使用。你的系统不再需要通知推流端停止推流，因为SRS会主动断开。你只需要开启如下配置：

# The timeout in seconds to disconnect publisher when idle, which means no players. # Note that 0 means no timeout or this feature is disabled. # Note that this feature conflicts with forward, because it disconnect the publisher stream. # Overwrite by env SRS_VHOST_PUBLISH_KICKOFF_FOR_IDLE for all vhosts. # default: 0 kickoff_for_idle 0;

详细过程可以参考这个PR。

Converting RTMP to HLS

如果需要将RTMP转HLS协议，请参考HLS.

Converting RTMP to HTTP-FLV

如果需要将RTMP转HTTP-FLV或HTTP-TS协议，请参考HTTP-FLV.

Converting RTMP to WebRTC

如果需要将RTMP转WebRTC协议，请参考WebRTC: RTMP to RTC.

Converting RTMP to MPEGTS-DASH

如果需要将RTMP转DASH协议是，参考DASH.

Converting SRT to RTMP

如果需要将SRT转RTMP协议，参考SRT.

Converting WebRTC to RTMP

如果需要将WebRTC协议转RTMP协议，参考WebRTC: RTC to RTMP.

RTMP Cluster

如果需要支持很多播放器播放，参考Edge Cluster.

如果需要支持很多推流，或者很多路流，参考Origin Cluster.

关于流媒体的负载均衡，还有很多其他方案，可以参考load balancing.

Low Latency RTMP

如果希望降低RTMP的延迟，请参考LowLatency.

Timestamp Jitter

SRS支持校准RTMP的时间戳，参考Jitter.

如果希望SRS能保持原始时间戳，参考ATC.

Performance

SRS使用writev实现高性能RTMP分发，参考benchmark