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WebRTC: Extract SrsRtcNetwork layer for UDP/TCP.
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winlin
parent
770d959148
commit
625069af7f
23 changed files with 698 additions and 528 deletions
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@ -116,6 +116,7 @@ auto_reload_for_docker on;
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# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
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# for example, 192.168.1.100:1935 10.10.10.100:1935
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# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
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# Overwrite by env SRS_LISTEN
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listen 1935;
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# the default chunk size is 128, max is 65536,
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# some client does not support chunk size change,
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@ -135,12 +136,14 @@ chunk_size 60000;
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# where the cli can only be used in shell/terminate.
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http_api {
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# whether http api is enabled.
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# Overwrite by env SRS_HTTP_API_ENABLED
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# default: off
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enabled on;
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# The http api listen entry is <[ip:]port>, For example, 192.168.1.100:8080, where the ip is optional, default to
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# 0.0.0.0, that is 8080 equals to 0.0.0.0:8080.
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# Note that you're able to use a dedicated port for HTTP API, such as 1985, to be different with HTTP server. In
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# this situation, you you must also set another HTTPS API port.
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# Overwrite by env SRS_HTTP_API_LISTEN
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# Default: 1985
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listen 8080;
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# whether enable crossdomain request.
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@ -164,11 +167,13 @@ http_api {
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# For https_api or HTTPS API.
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https {
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# Whether enable HTTPS API.
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# Overwrite by env SRS_HTTP_API_HTTPS_ENABLED
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# default: off
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enabled on;
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# The listen endpoint for HTTPS API.
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# Note that you're able to use a dedicated port for HTTPS API, such as 1990, and the HTTP API should not be
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# the same of HTTP server(8080) neither.
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# Overwrite by env SRS_HTTP_API_HTTPS_LISTEN
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# Default: 1990
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listen 8088;
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# The SSL private key file, generated by:
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@ -193,12 +198,14 @@ http_api {
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# need to open the feature http of vhost.
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http_server {
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# whether http streaming service is enabled.
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# Overwrite by env SRS_HTTP_SERVER_ENABLED
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# default: off
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enabled on;
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# the http streaming listen entry is <[ip:]port>
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# for example, 192.168.1.100:8080
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# where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
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# @remark, if use lower port, for instance 80, user must start srs by root.
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# Overwrite by env SRS_HTTP_SERVER_LISTEN
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# default: 8080
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listen 8080;
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# the default dir for http root.
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@ -211,9 +218,11 @@ http_server {
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# For https_server or HTTPS Streaming.
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https {
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# Whether enable HTTPS Streaming.
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# Overwrite by env SRS_HTTP_SERVER_HTTTPS_ENABLED
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# default: off
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enabled on;
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# The listen endpoint for HTTPS Streaming.
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# Overwrite by env SRS_HTTP_SERVER_HTTTPS_LISTEN
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# default: 8088
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listen 8088;
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# The SSL private key file, generated by:
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@ -297,9 +306,11 @@ vhost srt.vhost.srs.com {
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#############################################################################################
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rtc_server {
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# Whether enable WebRTC server.
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# Overwrite by env SRS_RTC_SERVER_ENABLED
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# default: off
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enabled on;
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# The udp listen port, we will reuse it for connections.
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# Overwrite by env SRS_RTC_SERVER_LISTEN
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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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@ -329,11 +340,13 @@ rtc_server {
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# x.x.x.x A specified IP address or DNS name, use * if 0.0.0.0.
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# @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
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# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
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# Overwrite by env SRS_RTC_SERVER_CANDIDATE
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# default: *
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candidate *;
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# If candidate is * or 0.0.0.0, means SRS could detect IP automatically, filtered by ip_family.
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# You can config this to off to disable the detecting, then SRS will try to parse the API hostname.
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# Note that browser might fail if no CANDIDATE specified.
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# Overwrite by env SRS_RTC_SERVER_USE_AUTO_DETECT_NETWORK_IP
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# Default: on
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use_auto_detect_network_ip on;
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# The IP family filter for auto discover candidate, it can be:
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@ -341,20 +354,24 @@ rtc_server {
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# ipv6 Filter IP v6 candidates.
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# all Filter all IP v4 or v6 candidates.
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# For example, if set to ipv4, we only use the IPv4 address as candidate.
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# Overwrite by env SRS_RTC_SERVER_IP_FAMILY
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# default: ipv4
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ip_family ipv4;
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# If api_as_candidates is on, SRS would try to use the IP of api server, specified by srs.sdk.js request:
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# api:string "http://r.ossrs.net:1985/rtc/v1/play/"
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# in this case, the r.ossrs.net and 39.107.238.185 will be added as candidates.
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# Overwrite by env SRS_RTC_SERVER_API_AS_CANDIDATES
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# Default: on
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api_as_candidates on;
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# If use api as CANDIDATE, whether resolve the api hostname.
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# Note that use original domain name as CANDIDATE, which might make Firefox failed, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
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# Note that if hostname is IPv4 address, always directly use it.
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# Overwrite by env SRS_RTC_SERVER_RESOLVE_API_DOMAIN
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# Default: on
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resolve_api_domain on;
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# If use api as CANDIDATE, whether keep original api domain name as CANDIDATE.
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# Note that use original domain name as CANDIDATE, which might make Firefox failed, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
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# Overwrite by env SRS_RTC_SERVER_KEEP_API_DOMAIN
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# Default: off
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keep_api_domain off;
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# Whether use ECDSA certificate.
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@ -390,6 +407,7 @@ rtc_server {
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vhost rtc.vhost.srs.com {
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rtc {
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# Whether enable WebRTC server.
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# Overwrite by env SRS_VHOST_RTC_ENABLED for all vhosts.
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# default: off
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enabled on;
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# Whether support NACK.
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@ -420,6 +438,7 @@ vhost rtc.vhost.srs.com {
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###############################################################
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# Whether enable transmuxing RTMP to RTC.
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# If enabled, transcode aac to opus.
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# Overwrite by env SRS_VHOST_RTC_RTMP_TO_RTC for all vhosts.
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# default: off
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rtmp_to_rtc off;
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# Whether keep B-frame, which is normal feature in live streaming,
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@ -428,6 +447,7 @@ vhost rtc.vhost.srs.com {
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keep_bframe off;
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###############################################################
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# Whether enable transmuxing RTC to RTMP.
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# Overwrite by env SRS_VHOST_RTC_RTC_TO_RTMP for all vhosts.
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# Default: off
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rtc_to_rtmp off;
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# The PLI interval in seconds, for RTC to RTMP.
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