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Code Issues Releases Wiki Activity

SquashSRS4: Support RTC2RTMP.

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This commit is contained in:
winlin 2021-05-01 22:15:57 +08:00
parent 0b62216999
commit 74bb47c13f
22 changed files with 1246 additions and 844 deletions
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101
trunk/conf/full.conf
View file

@ -446,14 +446,14 @@ srt_server {
rtc_server {
# Whether enable WebRTC server.
# default: off
enabled on;
enabled on;
# The udp listen port, we will reuse it for connections.
# default: 8000
listen 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
# $CANDIDATE Read the IP from ENV variable, use * if not set.
# 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.
@ -461,37 +461,38 @@ rtc_server {
# 192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
# And by multiple ENV variables:
# $CANDIDATE $EIP # TODO: Implements it.
# @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see also https://github.com/ossrs/srs/pull/1998#issuecomment-726442999
# @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
# default: *
candidate *;
# The IP family filter for candidate, it can be:
candidate *;
# The IP family filter for auto discover 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;
ip_family ipv4;
# Whether use ECDSA certificate.
# If not, use RSA certificate.
# default: on
ecdsa on;
ecdsa on;
# Whether encrypt RTP packet by SRTP.
# @remark Should always turn it on, or Chrome will fail.
# default: on
encrypt 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;
reuseport 1;
# Whether merge multiple NALUs into one.
# @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
# default: off
merge_nalus off;
merge_nalus off;
# Whether enable the perf stat at http://localhost:1985/api/v1/perf
# TODO: FIXME: We should enable it when refined.
# default: off
perf_stat off;
perf_stat off;
# For RTP packet and its payload cache.
rtp_cache {
# Whether enable the RTP packet cache.
@ -533,61 +534,67 @@ 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;
enabled on;
# Whether support NACK.
# default: on
nack on;
# Whether directly use the packet, avoid copy.
# default: on
nack_no_copy on;
# Whether support TWCC.
# default: on
twcc on;
# 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.
stun_timeout 30;
# The strict 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;
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;
###############################################################
# 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;
###############################################################
# For transmuxing RTC to RTMP.
# Whether trans-mux RTC to RTMP streaming.
# Default: off
rtc_to_rtmp off;
# The PLI interval in seconds, for RTC to RTMP.
# Note the available range is [0.5, 30]
# Default: 6.0
pli_for_rtmp 6.0;
}
# whether enable min delay mode for vhost.
###############################################################
# For transmuxing RTMP to RTC, it will impact the default values if RTC is on.
# Whether enable min delay mode for vhost.
# default: on, for RTC.
min_latency on;
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;
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;
# Whether directly use the packet, avoid copy.
# default: on
no_copy on;
}
# For TWCC.
twcc {
# Whether support TWCC.
# default: on
enabled on;
# default: 0 (For Real-Time, that is min_latency on)
# default: 1 (For WebRTC, that is min_latency off)
mw_msgs 0;
}
}

9
trunk/conf/rtc.conf
View file

@ -18,21 +18,20 @@ stats {
network 0;
}
rtc_server {
enabled on;
enabled on;
# Listen at udp://8000
listen 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;
# @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
bframe discard;
}
http_remux {
enabled on;

42
trunk/conf/rtc2rtmp.conf Normal file
View file

@ -0,0 +1,42 @@
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 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 $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
rtc_to_rtmp on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
}

47
trunk/conf/rtc_live.conf
View file

@ -1,47 +0,0 @@
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
}
http_api {
enabled on;
listen 1985;
}
rtc_server {
enabled on;
listen 8000;
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
bframe discard;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
tcp_nodelay on
min_latency on;
play {
gop_cache off;
queue_length 10;
mw_latency 100;
}
publish {
mr off;
}
}