排查SRS问题记录一（RTMP推流，RTC播放卡住）
作者：互联网
问题描述：
SRS Version=4.0.126. 配置文件配置：
listen 1935;
max_connections 100;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
http_api {
enabled on;
listen 1985;
}
stats {
network 0;
}
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/wiki/v4_CN_RTCWiki#config-candidate
candidate 192.168.71.35; #本机IP
}
vhost __defaultVhost__ {
rtc {
enabled on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
http_hooks {
enabled on;
on_rtc_play http://127.0.0.1/control/relay/subscriber;
}
}
注释：on_rtc_play http://127.0.0.1/control/relay/subscriber; 自定义开发，目的是在rtc播放拉流时，如果srs服务中没有直播流，可以通过回调到Nginx-RTMP触发RTMP推流到SRS中。
一、使用ffmpeg推流：
ffmpeg -re -i ~/Downloads/test.mp4 -c copy -f flv rtmp://127.0.0.1/live/livestream
二、使用http://127.0.0.1:8080/中SRS播放器播放。播放正常。
三、再打开一个页面使用http://127.0.0.1:8080/中SRS播放器播放，发现无法播放，并且第二部中的播放页面也会卡住。
只要开启rtc enabled on;即使不使用rtc播放，也存在如下问题
- 使用ffmpeg推流
- 使用ffplay播放流
- 使用ffmpeg推相同的流
- 第2步中的流可以播放，当时关闭播放再使用ffplay播放会失败。
问题分析
一、查看日志，发现有：
cleanup when unpublish
cleanup when unpublish, created=%u, deliver=%u
说明有触发推流断开。猜想可能是第二次进行RTC播放时，回调了http://127.0.0.1/control/relay/subscriber，使Nginx-RTMP服务又推流到SRS，并且app和name是相同的。这样第二次的RTMP推流由于是重复的流，所以会推流失败，然后进入unpublish调用链中。
二、在void SrsLiveSource::on_unpublish()函数中，将RTMP转RTC的bridger_进行了释放。
void SrsLiveSource::on_unpublish()
{
。。。。。略
if (bridger_) {
bridger_->on_unpublish();
srs_freep(bridger_);
}
。。。。。略
}
三、对于RTMP推流，是会经常出现重复推流的情况的。在Nginx-RTMP服务中，只会将第二次重复推流进行禁止，并不会影响第一次已经推上来的直播流。同理，SRS中的RTMP服务应该也有类似的逻辑。于是查看代码：
srs_error_t SrsRtmpConn::publishing(SrsLiveSource* source)
{
srs_error_t err = srs_success;
SrsRequest* req = info->req;
if (_srs_config->get_refer_enabled(req->vhost)) {
if ((err = refer->check(req->pageUrl, _srs_config->get_refer_publish(req->vhost))) != srs_success) {
return srs_error_wrap(err, "rtmp: referer check");
}
}
if ((err = http_hooks_on_publish()) != srs_success) {
return srs_error_wrap(err, "rtmp: callback on publish");
}
// TODO: FIXME: Should refine the state of publishing.
if ((err = acquire_publish(source)) == srs_success) {
// use isolate thread to recv,
// @see: https://github.com/ossrs/srs/issues/237
SrsPublishRecvThread rtrd(rtmp, req, srs_netfd_fileno(stfd), 0, this, source, _srs_context->get_id());
err = do_publishing(source, &rtrd);
rtrd.stop();
}
// whatever the acquire publish, always release publish.
// when the acquire error in the midlle-way, the publish state changed,
// but failed, so we must cleanup it.
// @see https://github.com/ossrs/srs/issues/474
// @remark when stream is busy, should never release it.
if (srs_error_code(err) != ERROR_SYSTEM_STREAM_BUSY) {
release_publish(source);
}
http_hooks_on_unpublish();
return err;
}
可以看到只有srs_error_code(err) != ERROR_SYSTEM_STREAM_BUSY才会将RTMP publish连接进行释放。
四、断点查看生成error code的代码
srs_error_t SrsRtmpConn::acquire_publish(SrsLiveSource* source)
{
srs_error_t err = srs_success;
SrsRequest* req = info->req;
// @see https://github.com/ossrs/srs/issues/2364
// Check whether GB28181 stream is busy.
#if defined(SRS_GB28181)
if (_srs_gb28181 != NULL) {
SrsGb28181RtmpMuxer* gb28181 = _srs_gb28181->fetch_rtmpmuxer(req->stream);
if (gb28181 != NULL) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "gb28181 stream %s busy", req->get_stream_url().c_str());
}
}
#endif
// Check whether RTC stream is busy.
#ifdef SRS_RTC
SrsRtcSource *rtc = NULL;
bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost);
if (rtc_server_enabled && rtc_enabled && !info->edge) {
if ((err = _srs_rtc_sources->fetch_or_create(req, &rtc)) != srs_success) {
return srs_error_wrap(err, "create source");
}
if (!rtc->can_publish()) {
return srs_error_new(ERROR_RTC_SOURCE_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
}
}
#endif
// Check whether RTMP stream is busy.
if (!source->can_publish(info->edge)) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp: stream %s is busy", req->get_stream_url().c_str());
}
// Bridge to RTC streaming.
#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
if (rtc) {
SrsRtcFromRtmpBridger *bridger = new SrsRtcFromRtmpBridger(rtc);
if ((err = bridger->initialize(req)) != srs_success) {
srs_freep(bridger);
return srs_error_wrap(err, "bridger init");
}
source->set_bridger(bridger);
}
#endif
// Start publisher now.
if (info->edge) {
return source->on_edge_start_publish();
} else {
return source->on_publish();
}
}
发现其在这一步直接返回错误了：
if (!rtc->can_publish()) {
return srs_error_new(ERROR_RTC_SOURCE_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
}
这是由于在第一个推流时，已经将RTMP转RTC的bridger_的SrsRtcSource中is_created_ = true;所以会返回RTC的推流已经存在。错误码并不是RTMP重复推流的错误码ERROR_SYSTEM_STREAM_BUSY。
问题修复
将acquire_publish中判断RTC推流重复和RTMP推流重复调换位置。如下：
srs_error_t SrsRtmpConn::acquire_publish(SrsLiveSource* source)
{
srs_error_t err = srs_success;
SrsRequest* req = info->req;
// @see https://github.com/ossrs/srs/issues/2364
// Check whether GB28181 stream is busy.
#if defined(SRS_GB28181)
if (_srs_gb28181 != NULL) {
SrsGb28181RtmpMuxer* gb28181 = _srs_gb28181->fetch_rtmpmuxer(req->stream);
if (gb28181 != NULL) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "gb28181 stream %s busy", req->get_stream_url().c_str());
}
}
#endif
// Check whether RTMP stream is busy.
if (!source->can_publish(info->edge)) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp: stream %s is busy", req->get_stream_url().c_str());
}
// Check whether RTC stream is busy.
#ifdef SRS_RTC
SrsRtcSource *rtc = NULL;
bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost);
if (rtc_server_enabled && rtc_enabled && !info->edge) {
if ((err = _srs_rtc_sources->fetch_or_create(req, &rtc)) != srs_success) {
return srs_error_wrap(err, "create source");
}
if (!rtc->can_publish()) {
return srs_error_new(ERROR_RTC_SOURCE_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
}
}
#endif
// Bridge to RTC streaming.
#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
if (rtc) {
SrsRtcFromRtmpBridger *bridger = new SrsRtcFromRtmpBridger(rtc);
if ((err = bridger->initialize(req)) != srs_success) {
srs_freep(bridger);
return srs_error_wrap(err, "bridger init");
}
source->set_bridger(bridger);
}
#endif
// Start publisher now.
if (info->edge) {
return source->on_edge_start_publish();
} else {
return source->on_publish();
}
}
标签：RTC,rtc,srs,req,publish,SRS,RTMP,error,stream 来源： https://blog.csdn.net/xiaofeilong89/article/details/118254711