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Squash: Fix bugs

This commit is contained in:
winlin 2022-01-13 18:26:28 +08:00
parent 4110fb14cb
commit 93aa0eb5ba
25 changed files with 198 additions and 184 deletions

View file

@ -24,7 +24,7 @@ function update_nav() {
}
// Special extra params, such as auth_key.
function user_extra_params(query, params) {
function user_extra_params(query, params, rtc) {
var queries = params || [];
for (var key in query.user_query) {
@ -90,6 +90,8 @@ function build_default_flv_url() {
function build_default_rtc_url(query) {
// The format for query string to overwrite configs of server.
console.log('?eip=x.x.x.x to overwrite candidate. 覆盖服务器candidate(外网IP)配置');
console.log('?api=x to overwrite WebRTC API(1985).');
console.log('?schema=http|https to overwrite WebRTC API protocol.');
var server = (!query.server)? window.location.hostname:query.server;
var vhost = (!query.vhost)? window.location.hostname:query.vhost;
@ -104,7 +106,7 @@ function build_default_rtc_url(query) {
if (query.schema && window.location.protocol !== query.schema + ':') {
queries.push('schema=' + query.schema);
}
queries = user_extra_params(query, queries);
queries = user_extra_params(query, queries, true);
var uri = "webrtc://" + server + api + "/" + app + "/" + stream + "?" + queries.join('&');
while (uri.lastIndexOf("?") === uri.length - 1) {

View file

@ -185,6 +185,12 @@ function SrsRtcPublisherAsync() {
var port = a.port;
if (!port) {
// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
}
// Guess by schema.
if (schema === 'http') {
port = 80;
} else if (schema === 'https') {
@ -267,6 +273,7 @@ function SrsRtcPlayerAsync() {
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// webrtc://r.ossrs.net:80/live/livestream
// or autostart the play:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
@ -413,6 +420,12 @@ function SrsRtcPlayerAsync() {
var port = a.port;
if (!port) {
// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
}
// Guess by schema.
if (schema === 'http') {
port = 80;
} else if (schema === 'https') {