support webrtc-datachannel sdp exchange

.github/workflows/release.yml

@@ -6,8 +6,9 @@ on:
     tags:
       - v6*
 
-# Declare default permissions as read only.
-permissions: read-all
+# For draft, need write permission.
+permissions:
+  contents: write
 
 jobs:
   envs:
@@ -284,6 +285,7 @@ jobs:
     needs:
       - envs
       - docker-srs
+      - test
     steps:
       ##################################################################################################################
       - name: Covert output to env

trunk/auto/depends.sh

@@ -549,7 +549,8 @@ if [[ $SRS_RTC == YES && $SRS_FFMPEG_OPUS != YES ]]; then
         rm -rf ${SRS_OBJS}/${SRS_PLATFORM}/opus-1.3.1 ${SRS_OBJS}/${SRS_PLATFORM}/3rdpatry/opus ${SRS_OBJS}/opus &&
         tar xf ${SRS_WORKDIR}/3rdparty/opus-1.3.1.tar.gz -C ${SRS_OBJS}/${SRS_PLATFORM} &&
         (
-            cd ${SRS_OBJS}/${SRS_PLATFORM}/opus-1.3.1 &&
+            # Opus requires automake 1.15, and fails for automake 1.16+, so we run autoreconf to fix it.
+            cd ${SRS_OBJS}/${SRS_PLATFORM}/opus-1.3.1 && autoreconf &&
             ./configure --prefix=${SRS_DEPENDS_LIBS}/${SRS_PLATFORM}/3rdpatry/opus --enable-static $OPUS_OPTIONS
         ) &&
         make -C ${SRS_OBJS}/${SRS_PLATFORM}/opus-1.3.1 ${SRS_JOBS} &&

trunk/conf/hevc.flv.conf

@@ -2,6 +2,10 @@ listen              1935;
 max_connections     1000;
 daemon              off;
 srs_log_tank        console;
+srt_server {
+    enabled on;
+    listen 10080;
+}
 http_api {
     enabled         on;
     listen          1985;
@@ -11,6 +15,10 @@ http_server {
     listen          8080;
 }
 vhost __defaultVhost__ {
+    srt {
+        enabled     on;
+        srt_to_rtmp on;
+    }
     http_remux {
         enabled     on;
         mount       [vhost]/[app]/[stream].flv;

trunk/conf/hevc.ts.conf

@@ -2,6 +2,10 @@ listen              1935;
 max_connections     1000;
 daemon              off;
 srs_log_tank        console;
+srt_server {
+    enabled on;
+    listen 10080;
+}
 http_api {
     enabled         on;
     listen          1985;
@@ -11,6 +15,10 @@ http_server {
     listen          8080;
 }
 vhost __defaultVhost__ {
+    srt {
+        enabled     on;
+        srt_to_rtmp on;
+    }
     http_remux {
         enabled     on;
         mount       [vhost]/[app]/[stream].ts;

trunk/conf/realtime.conf

@@ -7,7 +7,7 @@ max_connections     1000;
 daemon              off;
 srs_log_tank        console;
 vhost __defaultVhost__ {
-    tcp_nodelay     on
+    tcp_nodelay     on;
     min_latency     on;
 
     play {

trunk/conf/realtime.flv.conf

@@ -17,7 +17,7 @@ vhost __defaultVhost__ {
         mount       [vhost]/[app]/[stream].flv;
     }
 
-    tcp_nodelay     on
+    tcp_nodelay     on;
     min_latency     on;
 
     play {

trunk/doc/CHANGELOG.md

@@ -8,6 +8,11 @@ The changelog for SRS.
 
 ## SRS 6.0 Changelog
 
+* v6.0, 2023-03-27, Merge [#3450](https://github.com/ossrs/srs/pull/3450): WebRTC: Error message carries the SDP when failed. v6.0.39 (#3450)
+* v6.0, 2023-03-25, Merge [#3477](https://github.com/ossrs/srs/pull/3477): Remove unneccessary NULL check in srs_freep. v6.0.38 (#3477)
+* v6.0, 2023-03-25, Merge [#3455](https://github.com/ossrs/srs/pull/3455): RTC: Call on_play before create session, for it might be freed for timeout. v6.0.37 (#3455)
+* v6.0, 2023-03-22, Merge [#3427](https://github.com/ossrs/srs/pull/3427): WHIP: Support DELETE resource for Larix Broadcaster. v6.0.36 (#3427)
+* v6.0, 2023-03-20, Merge [#3460](https://github.com/ossrs/srs/pull/3460): WebRTC: Support WHIP/WHEP players. v6.0.35 (#3460)
 * v6.0, 2023-03-07, Merge [#3441](https://github.com/ossrs/srs/pull/3441): HEVC: webrtc support hevc on safari. v6.0.34 (#3441)
 * v6.0, 2023-03-07, Merge [#3446](https://github.com/ossrs/srs/pull/3446): WebRTC: Warning if no ideal profile. v6.0.33 (#3446)
 * v6.0, 2023-03-06, Merge [#3445](https://github.com/ossrs/srs/pull/3445): Support configure for generic linux. v6.0.32 (#3445)
@@ -48,6 +53,11 @@ The changelog for SRS.
 
 ## SRS 5.0 Changelog
 
+* v5.0, 2023-03-27, Merge [#3450](https://github.com/ossrs/srs/pull/3450): WebRTC: Error message carries the SDP when failed. v5.0.151 (#3450)
+* v5.0, 2023-03-25, Merge [#3477](https://github.com/ossrs/srs/pull/3477): Remove unneccessary NULL check in srs_freep. v5.0.150 (#3477)
+* v5.0, 2023-03-25, Merge [#3455](https://github.com/ossrs/srs/pull/3455): RTC: Call on_play before create session, for it might be freed for timeout. v5.0.149 (#3455)
+* v5.0, 2023-03-22, Merge [#3427](https://github.com/ossrs/srs/pull/3427): WHIP: Support DELETE resource for Larix Broadcaster. v5.0.148 (#3427)
+* v5.0, 2023-03-20, Merge [#3460](https://github.com/ossrs/srs/pull/3460): WebRTC: Support WHIP/WHEP players. v5.0.147 (#3460)
 * v5.0, 2023-03-07, Merge [#3446](https://github.com/ossrs/srs/pull/3446): WebRTC: Warning if no ideal profile. v5.0.146 (#3446)
 * v5.0, 2023-03-06, Merge [#3445](https://github.com/ossrs/srs/pull/3445): Support configure for generic linux. v5.0.145 (#3445)
 * v5.0, 2023-03-04, Merge [#3105](https://github.com/ossrs/srs/pull/3105): Kickoff publisher when stream is idle, which means no players. v5.0.144 (#3105)

trunk/doc/Features.md

@@ -61,6 +61,7 @@ The features of SRS.
 - [x] RTC: [Experimental] Support transmux RTC to RTMP, [#2093](https://github.com/ossrs/srs/issues/2093). v4.0.95
 - [x] RTC: [Experimental] Support WebRTC over TCP directly, [#2852](https://github.com/ossrs/srs/issues/2852). v5.0.60+
 - [x] RTC: [Experimental] Support WHIP(WebRTC-HTTP ingestion protocol), [#3170](https://github.com/ossrs/srs/issues/3170). v5.0.61+
+- [x] RTC: [Experimental] Support [Larix Broadcaster](https://softvelum.com/larix/), [#3476](https://github.com/ossrs/srs/issues/3476). v5.0.148+
 - [x] Other: Support ingesting([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/ingest), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/ingest)) other protocols to SRS by FFMPEG. v1.0.0+
 - [x] Other: Support forwarding([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/forward), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/forward)) to other RTMP servers. v1.0.0+
 - [x] Other: Support transcoding([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/ffmpeg), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/ffmpeg)) by FFMPEG. v1.0.0+
@@ -74,6 +75,7 @@ The features of SRS.
 - [x] Other: [Experimental] Support pushing MPEG-TS over UDP, please read [bug #250](https://github.com/ossrs/srs/issues/250). v2.0.111+
 - [x] Other: [Experimental] Support pushing FLV over HTTP POST, please read wiki([CN](https://ossrs.net/lts/zh-cn/docs/v4/doc/streamer#push-http-flv-to-srs), [EN](https://ossrs.io/lts/en-us/docs/v4/doc/streamer#push-http-flv-to-srs)). v2.0.163+
 - [x] Other: [Experimental] Support push stream by GB28181, [#3176](https://github.com/ossrs/srs/issues/3176). v5.0.74+
+- [x] Other: Support WHIP/WHEP player, [#3460](https://github.com/ossrs/srs/pull/3460). v5.0.147+
 - [ ] System: Proxy to extend origin servers, [#3138](https://github.com/ossrs/srs/issues/3138).
 - [ ] System: Support source cleanup for idle streams, [#413](https://github.com/ossrs/srs/issues/413).
 - [ ] System: Support JT808 and JT1708 for transport, [#3420](https://github.com/ossrs/srs/issues/3420).

trunk/research/players/js/srs.page.js

@@ -17,6 +17,8 @@ function update_nav() {
     $("#nav_srs_player").attr("href", "srs_player.html" + window.location.search);
     $("#nav_rtc_player").attr("href", "rtc_player.html" + window.location.search);
     $("#nav_rtc_publisher").attr("href", "rtc_publisher.html" + window.location.search);
+    $("#nav_whip").attr("href", "whip.html" + window.location.search);
+    $("#nav_whep").attr("href", "whep.html" + window.location.search);
     $("#nav_srs_publisher").attr("href", "srs_publisher.html" + window.location.search);
     $("#nav_srs_chat").attr("href", "srs_chat.html" + window.location.search);
     $("#nav_srs_bwt").attr("href", "srs_bwt.html" + window.location.search);
@@ -116,6 +118,38 @@ function build_default_rtc_url(query) {
     return uri;
 };
 
+function build_default_whip_whep_url(query, apiPath) {
+    // 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;
+    var app = (!query.app)? "live":query.app;
+    var stream = (!query.stream)? "livestream":query.stream;
+    var api = ':' + (query.api || (window.location.protocol === 'http:' ? '1985' : '1990'));
+
+    var queries = [];
+    if (server !== vhost && vhost !== "__defaultVhost__") {
+        queries.push("vhost=" + vhost);
+    }
+    if (query.schema && window.location.protocol !== query.schema + ':') {
+        queries.push('schema=' + query.schema);
+    }
+    queries = user_extra_params(query, queries, true);
+
+    var uri = window.location.protocol + "//" + server + api + apiPath + "?app=" + app + "&stream=" + stream + "&" + queries.join('&');
+    while (uri.lastIndexOf("?") === uri.length - 1) {
+        uri = uri.slice(0, uri.length - 1);
+    }
+    while (uri.lastIndexOf("&") === uri.length - 1) {
+        uri = uri.slice(0, uri.length - 1);
+    }
+
+    return uri;
+}
+
 /**
 * initialize the page.
 * @param flv_url the div id contains the flv stream url to play
@@ -136,3 +170,11 @@ function srs_init_rtc(id, query) {
     update_nav();
     $(id).val(build_default_rtc_url(query));
 }
+function srs_init_whip(id, query) {
+    update_nav();
+    $(id).val(build_default_whip_whep_url(query, '/rtc/v1/whip/'));
+}
+function srs_init_whep(id, query) {
+    update_nav();
+    $(id).val(build_default_whip_whep_url(query, '/rtc/v1/whip-play/'));
+}

trunk/research/players/js/srs.sdk.js

@@ -83,7 +83,7 @@ function SrsRtcPublisherAsync() {
             const xhr = new XMLHttpRequest();
             xhr.onload = function() {
                 if (xhr.readyState !== xhr.DONE) return;
-                if (xhr.status !== 200) return reject(xhr);
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                 const data = JSON.parse(xhr.responseText);
                 console.log("Got answer: ", data);
                 return data.code ? reject(xhr) : resolve(data);
@@ -134,14 +134,14 @@ function SrsRtcPublisherAsync() {
                 api += '/';
             }
 
-            apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+            var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
             for (var key in urlObject.user_query) {
                 if (key !== 'api' && key !== 'play') {
                     apiUrl += '&' + key + '=' + urlObject.user_query[key];
                 }
             }
             // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
-            var apiUrl = apiUrl.replace(api + '&', api + '?');
+            apiUrl = apiUrl.replace(api + '&', api + '?');
 
             var streamUrl = urlObject.url;
 
@@ -318,7 +318,7 @@ function SrsRtcPlayerAsync() {
             const xhr = new XMLHttpRequest();
             xhr.onload = function() {
                 if (xhr.readyState !== xhr.DONE) return;
-                if (xhr.status !== 200) return reject(xhr);
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
                 const data = JSON.parse(xhr.responseText);
                 console.log("Got answer: ", data);
                 return data.code ? reject(xhr) : resolve(data);
@@ -369,14 +369,14 @@ function SrsRtcPlayerAsync() {
                 api += '/';
             }
 
-            apiUrl = schema + '//' + urlObject.server + ':' + port + api;
+            var apiUrl = schema + '//' + urlObject.server + ':' + port + api;
             for (var key in urlObject.user_query) {
                 if (key !== 'api' && key !== 'play') {
                     apiUrl += '&' + key + '=' + urlObject.user_query[key];
                 }
             }
             // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
-            var apiUrl = apiUrl.replace(api + '&', api + '?');
+            apiUrl = apiUrl.replace(api + '&', api + '?');
 
             var streamUrl = urlObject.url;
 
@@ -510,6 +510,145 @@ function SrsRtcPlayerAsync() {
     return self;
 }
 
+// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
+// Async-awat-prmise based SRS RTC Publisher by WHIP.
+function SrsRtcWhipWhepAsync() {
+    var self = {};
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
+    self.constraints = {
+        audio: true,
+        video: {
+            width: {ideal: 320, max: 576}
+        }
+    };
+
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+    // @url The WebRTC url to publish with, for example:
+    //      http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
+    self.publish = async function (url) {
+        if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
+
+        self.pc.addTransceiver("audio", {direction: "sendonly"});
+        self.pc.addTransceiver("video", {direction: "sendonly"});
+
+        if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
+            throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
+        }
+        var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+        stream.getTracks().forEach(function (track) {
+            self.pc.addTrack(track);
+
+            // Notify about local track when stream is ok.
+            self.ontrack && self.ontrack({track: track});
+        });
+
+        var offer = await self.pc.createOffer();
+        await self.pc.setLocalDescription(offer);
+        const answer = await new Promise(function (resolve, reject) {
+            console.log("Generated offer: ", offer);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+                const data = xhr.responseText;
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', url, true);
+            xhr.setRequestHeader('Content-type', 'application/sdp');
+            xhr.send(offer.sdp);
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: answer})
+        );
+
+        return self.__internal.parseId(url, offer.sdp, answer);
+    };
+
+    // See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+    // @url The WebRTC url to play with, for example:
+    //      http://localhost:1985/rtc/v1/whip-play/?app=live&stream=livestream
+    self.play = async function(url) {
+        if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
+
+        self.pc.addTransceiver("audio", {direction: "recvonly"});
+        self.pc.addTransceiver("video", {direction: "recvonly"});
+
+        var offer = await self.pc.createOffer();
+        await self.pc.setLocalDescription(offer);
+        const answer = await new Promise(function(resolve, reject) {
+            console.log("Generated offer: ", offer);
+
+            const xhr = new XMLHttpRequest();
+            xhr.onload = function() {
+                if (xhr.readyState !== xhr.DONE) return;
+                if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
+                const data = xhr.responseText;
+                console.log("Got answer: ", data);
+                return data.code ? reject(xhr) : resolve(data);
+            }
+            xhr.open('POST', url, true);
+            xhr.setRequestHeader('Content-type', 'application/sdp');
+            xhr.send(offer.sdp);
+        });
+        await self.pc.setRemoteDescription(
+            new RTCSessionDescription({type: 'answer', sdp: answer})
+        );
+
+        return self.__internal.parseId(url, offer.sdp, answer);
+    };
+
+    // Close the publisher.
+    self.close = function () {
+        self.pc && self.pc.close();
+        self.pc = null;
+    };
+
+    // The callback when got local stream.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    self.ontrack = function (event) {
+        // Add track to stream of SDK.
+        self.stream.addTrack(event.track);
+    };
+
+    self.pc = new RTCPeerConnection(null);
+
+    // To keep api consistent between player and publisher.
+    // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+    // @see https://webrtc.org/getting-started/media-devices
+    self.stream = new MediaStream();
+
+    // Internal APIs.
+    self.__internal = {
+        parseId: (url, offer, answer) => {
+            let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
+            sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
+            sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
+            sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
+
+            const a = document.createElement("a");
+            a.href = url;
+            return {
+                sessionid: sessionid, // Should be ice-ufrag of answer:offer.
+                simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
+            };
+        },
+    };
+
+    // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
+    self.pc.ontrack = function(event) {
+        if (self.ontrack) {
+            self.ontrack(event);
+        }
+    };
+
+    return self;
+}
+
 // Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
 // https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
 function SrsRtcFormatSenders(senders, kind) {