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WebRTC: Support config, listener and SDP for TCP transport.
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14 changed files with 767 additions and 386 deletions
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@ -302,6 +302,27 @@ rtc_server {
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# The udp listen port, we will reuse it for connections.
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# default: 8000
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listen 8000;
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# For WebRTC over TCP directly, not TURN, see https://github.com/ossrs/srs/issues/2852
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# Some network does not support UDP, or not very well, so we use TCP like HTTP/80 port for firewall traversing.
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tcp {
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# Whether enable WebRTC over TCP.
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# Overwrite by env SRS_RTC_SERVER_TCP_ENABLED
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# Default: off
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enabled off;
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# The TCP listen port for WebRTC. Highly recommend is some normally used ports, such as TCP/80, TCP/443,
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# TCP/8000, TCP/8080 etc. However SRS default to TCP/8000 corresponding to UDP/8000.
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# Overwrite by env SRS_RTC_SERVER_TCP_LISTEN
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# Default: 8000
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listen 8000;
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}
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# The protocol for candidate to use, it can be:
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# udp Generate UDP candidates. Note that UDP server is always enabled for WebRTC.
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# tcp Generate TCP candidates. Fail if rtc_server.tcp(WebRTC over TCP) is disabled.
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# all Generate UDP+TCP candidates. Ignore if rtc_server.tcp(WebRTC over TCP) is disabled.
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# Note that if both are connected, we will use the first connected(DTLS done) one.
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# Overwrite by env SRS_RTC_SERVER_PROTOCOL
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# Default: udp
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protocol udp;
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# The exposed candidate IPs, response in SDP candidate line. It can be:
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# * Retrieve server IP automatically, from all network interfaces.
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# $CANDIDATE Read the IP from ENV variable, use * if not set.
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