mirror of
https://github.com/ossrs/srs.git
synced 2025-03-09 15:49:59 +00:00
Merge branch '4.0release' into merge/develop
This commit is contained in:
commit
d30145d500
32 changed files with 9977 additions and 131 deletions
|
@ -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;
|
||||
}
|
||||
|
||||
|
|
|
@ -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;
|
||||
|
|
|
@ -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;
|
||||
}
|
||||
}
|
||||
|
|
|
@ -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;
|
||||
}
|
||||
|
|
|
@ -44,7 +44,7 @@ rtc_server {
|
|||
vhost __defaultVhost__ {
|
||||
rtc {
|
||||
enabled on;
|
||||
bframe discard;
|
||||
keep_bframe off;
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
@ -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 {
|
||||
|
|
|
@ -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;
|
||||
|
|
|
@ -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
40
trunk/conf/rtmp2rtc.conf
Normal file
|
@ -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;
|
||||
}
|
||||
}
|
||||
|
|
@ -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;
|
||||
}
|
||||
}
|
||||
|
|
|
@ -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;
|
||||
|
|
Loading…
Add table
Add a link
Reference in a new issue