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Change GB28181 to feature/gb28181. 4.0.127

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winlin 2021-06-16 08:27:37 +08:00
parent 4e93696bc6
commit 68c48e27f5
35 changed files with 14 additions and 9868 deletions
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83
trunk/conf/full.conf
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@ -366,89 +366,6 @@ stream_caster {
listen 8936;
}
# GB28181
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# gb28181, Push GB28181 to SRS.
caster gb28181;
# the output rtmp url.
# for gb28181 caster, the typically output url:
# rtmp://127.0.0.1/live/[stream]
# where the [stream] is the VideoChannelCodecID.
output rtmp://127.0.0.1/live/[stream];
# the listen port for stream caster.
# for gb28181 caster, listen at udp port. for example, 9000.
# @remark We can bundle all gb28181 to this port, to reuse this port.
# User can choose to bundle port in API port_mode or SIP invite_port_fixed.
listen 9000;
# Listen as TCP if on; otherwise, listen as UDP.
# default: off
tcp_enable off;
# If not bundle ports, use specified ports for each stream.
rtp_port_min 58200;
rtp_port_max 58300;
# Whether wait for keyframe then forward to RTMP.
# default: on
wait_keyframe on;
# Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
# default: 30
rtp_idle_timeout 30;
# Whether has audio.
# @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
# default: off
audio_enable off;
# The exposed IP to receive media stream.
# * 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 $EIP, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
# 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.
# Also by IP or DNS names:
# 192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
# And by multiple ENV variables:
# $CANDIDATE $EIP # TODO: Implements it.
# default: *
host *;
#The media channel is automatically created according to the received RTP packet,
# and the channel ID is generated according to the RTP SSRC
# channelid format: 'chid[ssrc]' [ssrc] is rtp's ssrc
auto_create_channel off;
sip {
# Whether enable embeded SIP server.
# default: on
enabled on;
# The SIP listen port.
# default: 5060
listen 5060;
# The SIP server ID.
# default: 34020000002000000001
serial 34020000002000000001;
# The SIP server domain.
# default: 3402000000
realm 3402000000;
# The SIP ACK response timeout in seconds.
# default: 30
ack_timeout 30;
# The keepalive timeout in seconds.
# default: 120
keepalive_timeout 120;
# Whether play immediately after registered.
# default: on
auto_play on;
# Whether bundle media stream port.
# default: on
invite_port_fixed on;
# interval to query equipment list from equipment or subordinate domain, unit(s)
# default: 60
query_catalog_interval 60;
}
}
#############################################################################################
# SRT server section
#############################################################################################