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Merge branch '4.0release' into merge/develop

This commit is contained in:
winlin 2021-10-12 08:34:22 +08:00
commit d30145d500
32 changed files with 9977 additions and 131 deletions

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@ -25,7 +25,7 @@ rtc_server {
# 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
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}

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@ -25,13 +25,17 @@ rtc_server {
# 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
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp on;
}
http_remux {
enabled on;

View file

@ -17,6 +17,7 @@ http_server {
rtc_server {
enabled on;
listen 8000;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
@ -29,5 +30,9 @@ vhost __defaultVhost__ {
}
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
}

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@ -388,7 +388,7 @@ rtc_server {
# And by multiple ENV variables:
# $CANDIDATE $EIP # TODO: Implements it.
# @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name.
# @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
# default: *
candidate *;
# The IP family filter for auto discover candidate, it can be:
@ -459,19 +459,16 @@ vhost rtc.vhost.srs.com {
# default: 0
drop_for_pt 0;
###############################################################
# For transmuxing RTMP to RTC, the strategy for bframe.
# keep Keep bframe, which may make browser with playing problems.
# discard Discard bframe, maybe cause browser with little problems.
# default: discard
bframe discard;
# For transmuxing RTMP to RTC, the strategy for aac audio.
# transcode Transcode aac to opus.
# discard Discard aac audio packet.
# default: transcode
aac transcode;
# Whether enable transmuxing RTMP to RTC.
# If enabled, transcode aac to opus.
# default: off
rtmp_to_rtc off;
# Whether keep B-frame, which is normal feature in live streaming,
# but usually disabled in RTC.
# default: off
keep_bframe off;
###############################################################
# For transmuxing RTC to RTMP.
# Whether trans-mux RTC to RTMP streaming.
# Whether enable transmuxing RTC to RTMP.
# Default: off
rtc_to_rtmp off;
# The PLI interval in seconds, for RTC to RTMP.
@ -889,6 +886,7 @@ vhost publish.srs.com {
# 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.
# default: on
parse_sps on;
}

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@ -44,7 +44,7 @@ rtc_server {
vhost __defaultVhost__ {
rtc {
enabled on;
bframe discard;
keep_bframe off;
}
}

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@ -20,20 +20,15 @@ stats {
}
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
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
bframe discard;
rtmp_to_rtc on;
keep_bframe off;
rtc_to_rtmp on;
}
play {

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@ -21,19 +21,18 @@ stats {
}
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
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
http_remux {
enabled on;

View file

@ -19,19 +19,17 @@ stats {
}
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
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp on;
}
http_remux {

40
trunk/conf/rtmp2rtc.conf Normal file
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@ -0,0 +1,40 @@
listen 1935;
max_connections 1000;
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 8000;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
}

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@ -18,6 +18,7 @@ http_server {
rtc_server {
enabled on;
listen 8000;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
@ -30,5 +31,9 @@ vhost __defaultVhost__ {
}
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
}

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@ -32,7 +32,7 @@ rtc_server {
# 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
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
@ -40,6 +40,10 @@ rtc_server {
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
http_remux {
enabled on;