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WebRTC: Support config, listener and SDP for TCP transport.

This commit is contained in:
winlin 2022-09-02 22:39:19 +08:00
parent 424713a31d
commit 770d959148
14 changed files with 767 additions and 386 deletions

View file

@ -302,6 +302,27 @@ rtc_server {
# The udp listen port, we will reuse it for connections.
# default: 8000
listen 8000;
# For WebRTC over TCP directly, not TURN, see https://github.com/ossrs/srs/issues/2852
# Some network does not support UDP, or not very well, so we use TCP like HTTP/80 port for firewall traversing.
tcp {
# Whether enable WebRTC over TCP.
# Overwrite by env SRS_RTC_SERVER_TCP_ENABLED
# Default: off
enabled off;
# The TCP listen port for WebRTC. Highly recommend is some normally used ports, such as TCP/80, TCP/443,
# TCP/8000, TCP/8080 etc. However SRS default to TCP/8000 corresponding to UDP/8000.
# Overwrite by env SRS_RTC_SERVER_TCP_LISTEN
# Default: 8000
listen 8000;
}
# The protocol for candidate to use, it can be:
# udp Generate UDP candidates. Note that UDP server is always enabled for WebRTC.
# tcp Generate TCP candidates. Fail if rtc_server.tcp(WebRTC over TCP) is disabled.
# all Generate UDP+TCP candidates. Ignore if rtc_server.tcp(WebRTC over TCP) is disabled.
# Note that if both are connected, we will use the first connected(DTLS done) one.
# Overwrite by env SRS_RTC_SERVER_PROTOCOL
# Default: udp
protocol udp;
# The exposed candidate IPs, response in SDP candidate line. It can be:
# * Retrieve server IP automatically, from all network interfaces.
# $CANDIDATE Read the IP from ENV variable, use * if not set.