GB28181: Support external SIP server. v6.0.144 (#4101)

For #3369 to support an external powerful SIP server, do not use the
embedded SIP server of SRS.
For more information, detailed steps, system architecture, and
background explanation, please see
https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip

---------

Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>

trunk/conf/full.conf

@@ -676,9 +676,11 @@ stream_caster {
     listen 9000;
     # SIP server for GB28181. Please note that this is only a demonstrated SIP server, please never use it in your
     # online production environment. Instead please use [jsip](https://github.com/usnistgov/jsip) and there is a demo
-    # [srs-sip](https://github.com/ossrs/srs-sip) also base on it.
+    # [srs-sip](https://github.com/ossrs/srs-sip) also base on it, for more information please see project
+    # [GB: External SIP](https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip).
     sip {
-        # Whether enable embedded SIP server.
+        # Whether enable embedded SIP server. Please disable it if you want to use your own SIP server, see
+        # [GB: External SIP](https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip).
         # Default: on
         enabled on;
         # The SIP listen port, for TCP protocol.

trunk/conf/gb28181-without-sip.conf

@@ -0,0 +1,53 @@
+
+listen 1935;
+max_connections 1000;
+daemon off;
+srs_log_tank console;
+
+stream_caster {
+    enabled on;
+    caster gb28181;
+    output rtmp://127.0.0.1/live/[stream];
+    listen 9000;
+    sip {
+        enabled off;
+    }
+}
+
+http_server {
+    enabled on;
+    listen 8080;
+    dir ./objs/nginx/html;
+}
+
+http_api {
+    enabled on;
+    listen 1985;
+}
+stats {
+    network 0;
+}
+rtc_server {
+    enabled on;
+    listen 8000; # UDP port
+    # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
+    candidate $CANDIDATE;
+}
+
+vhost __defaultVhost__ {
+    rtc {
+        enabled on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
+        rtmp_to_rtc on;
+        # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
+        rtc_to_rtmp on;
+    }
+    http_remux {
+        enabled on;
+        mount [vhost]/[app]/[stream].flv;
+    }
+    hls {
+        enabled on;
+    }
+}
+

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v6-changes"></a>
 
 ## SRS 6.0 Changelog
+* v6.0, 2024-07-27, Merge [#4101](https://github.com/ossrs/srs/pull/4101): GB28181: Support external SIP server. v6.0.144 (#4101)
 * v6.0, 2024-07-24, Merge [#4115](https://github.com/ossrs/srs/pull/4115): HLS: Add missing newline to end of session manifest. v6.0.143 (#4115)
 * v6.0, 2024-07-24, Merge [#4029](https://github.com/ossrs/srs/pull/4029): Player: Fix empty img tag occupy 20px size in safari. v6.0.142 (#4029)
 * v6.0, 2024-07-24, Merge [#4063](https://github.com/ossrs/srs/pull/4063): let http-remux ts stream support guess_has_av feature;. v6.0.141 (#4063)

trunk/src/app/srs_app_gb28181.cpp

@@ -22,6 +22,10 @@
 #include <srs_app_pithy_print.hpp>
 #include <srs_app_rtmp_conn.hpp>
 #include <srs_protocol_raw_avc.hpp>
+#include <srs_app_server.hpp>
+#include <srs_protocol_json.hpp>
+#include <srs_app_http_api.hpp>
+#include <srs_app_statistic.hpp>
 
 #include <sstream>
 using namespace std;
@@ -420,12 +424,12 @@ srs_error_t SrsGbListener::initialize(SrsConfDirective* conf)
 
     bool sip_enabled = _srs_config->get_stream_caster_sip_enable(conf);
     if (!sip_enabled) {
-        return srs_error_new(ERROR_GB_CONFIG, "GB SIP is required");
+        srs_warn("GB SIP is disabled.");
+    } else {
+        int port = _srs_config->get_stream_caster_sip_listen(conf);
+        sip_listener_->set_endpoint(ip, port)->set_label("SIP-TCP");
     }
 
-    int port = _srs_config->get_stream_caster_sip_listen(conf);
-    sip_listener_->set_endpoint(ip, port)->set_label("SIP-TCP");
-
     return err;
 }
 
@@ -441,6 +445,24 @@ srs_error_t SrsGbListener::listen()
         return srs_error_wrap(err, "listen");
     }
 
+    if ((err = listen_api()) != srs_success) {
+        return srs_error_wrap(err, "listen api");
+    }
+
+    return err;
+}
+
+srs_error_t SrsGbListener::listen_api()
+{
+    srs_error_t err = srs_success;
+
+    // TODO: FIXME: Fetch api from hybrid manager, not from SRS.
+    ISrsHttpServeMux* http_api_mux = _srs_hybrid->srs()->instance()->api_server();
+
+    if ((err = http_api_mux->handle("/gb/v1/publish/", new SrsGoApiGbPublish(conf_))) != srs_success) {
+        return srs_error_wrap(err, "handle publish");
+    }
+
     return err;
 }
 
@@ -549,11 +571,16 @@ std::string SrsGbSipTcpConn::device_id()
     return register_->device_id();
 }
 
+void SrsGbSipTcpConn::set_device_id(const std::string &id)
+{
+    register_->from_address_user_ = id;
+}
+
 void SrsGbSipTcpConn::set_cid(const SrsContextId& cid)
 {
     if (owner_cid_) owner_cid_->set_cid(cid);
-    receiver_->set_cid(cid);
-    sender_->set_cid(cid);
+    if (receiver_) receiver_->set_cid(cid);
+    if (sender_) sender_->set_cid(cid);
     cid_ = cid;
 }
 
@@ -2681,5 +2708,117 @@ void srs_sip_parse_address(const std::string& address, std::string& user, std::s
     }
 }
 
+SrsGoApiGbPublish::SrsGoApiGbPublish(SrsConfDirective* conf)
+{
+    conf_ = conf->copy();
+}
+
+SrsGoApiGbPublish::~SrsGoApiGbPublish()
+{
+    srs_freep(conf_);
+}
+
+srs_error_t SrsGoApiGbPublish::serve_http(ISrsHttpResponseWriter *w, ISrsHttpMessage *r)
+{
+    srs_error_t err = srs_success;
+
+    SrsUniquePtr<SrsJsonObject> res(SrsJsonAny::object());
+
+    if ((err = do_serve_http(w, r, res.get())) != srs_success) {
+        srs_warn("GB error %s", srs_error_desc(err).c_str());
+        res->set("code", SrsJsonAny::integer(srs_error_code(err)));
+        res->set("desc", SrsJsonAny::str(srs_error_code_str(err).c_str()));
+        srs_freep(err);
+        return srs_api_response(w, r, res->dumps());
+    }
+
+    return srs_api_response(w, r, res->dumps());
+}
+
+srs_error_t SrsGoApiGbPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res)
+{
+    srs_error_t err = srs_success;
+
+    // For each GB session, we use short-term HTTP connection.
+    w->header()->set("Connection", "Close");
+
+    // Parse req, the request json object, from body.
+    SrsSharedPtr<SrsJsonObject> req;
+    if (true) {
+        string req_json;
+        if ((err = r->body_read_all(req_json)) != srs_success) {
+            return srs_error_wrap(err, "read body");
+        }
+
+        SrsJsonAny* json = SrsJsonAny::loads(req_json);
+        if (!json || !json->is_object()) {
+            srs_freep(json);
+            return srs_error_new(ERROR_HTTP_DATA_INVALID, "invalid body %s", req_json.c_str());
+        }
+
+        req = SrsSharedPtr<SrsJsonObject>(json->to_object());
+    }
+
+    // Fetch params from req object.
+    SrsJsonAny* prop = NULL;
+    if ((prop = req->ensure_property_string("id")) == NULL) {
+        return srs_error_new(ERROR_HTTP_DATA_INVALID, "id required");
+    }
+    string id = prop->to_str();
+
+    if ((prop = req->ensure_property_string("ssrc")) == NULL) {
+        return srs_error_new(ERROR_HTTP_DATA_INVALID, "ssrc required");
+    }
+    uint64_t ssrc = atoi(prop->to_str().c_str());
+
+    if ((err = bind_session(id, ssrc)) != srs_success) {
+        return srs_error_wrap(err, "bind session");
+    }
+
+    res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
+    int port = _srs_config->get_stream_caster_listen(conf_);
+    res->set("port", SrsJsonAny::integer(port));
+    res->set("is_tcp", SrsJsonAny::boolean(true)); // only tcp supported
+
+    srs_trace("GB publish id: %s, ssrc=%lu", id.c_str(), ssrc);
+
+    return err;
+}
+
+srs_error_t SrsGoApiGbPublish::bind_session(std::string id, uint64_t ssrc)
+{
+    srs_error_t err = srs_success;
+
+    SrsSharedResource<SrsGbSession>* session = NULL;
+    session = dynamic_cast<SrsSharedResource<SrsGbSession>*>(_srs_gb_manager->find_by_id(id));
+    if (session) {
+        return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "stream already exists");
+    }
+
+    session = dynamic_cast<SrsSharedResource<SrsGbSession>*>(_srs_gb_manager->find_by_fast_id(ssrc));
+    if (session) {
+        return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "ssrc already exists");
+    }
+
+    // Create new GB session.
+    SrsGbSession* raw_session = new SrsGbSession();
+    raw_session->setup(conf_);
+
+    session = new SrsSharedResource<SrsGbSession>(raw_session);
+    _srs_gb_manager->add_with_id(id, session);
+    _srs_gb_manager->add_with_fast_id(ssrc, session);
+
+    SrsExecutorCoroutine* executor = new SrsExecutorCoroutine(_srs_gb_manager, session, raw_session, raw_session);
+    raw_session->setup_owner(session, executor, executor);
+    raw_session->sip_transport()->set_device_id(id);
+
+    if ((err = executor->start()) != srs_success) {
+        srs_freep(executor);
+        return srs_error_wrap(err, "gb session");
+    }
+    
+    return err;
+}
+
 SrsResourceManager* _srs_gb_manager = NULL;
 

trunk/src/app/srs_app_gb28181.hpp

@@ -89,6 +89,34 @@ enum SrsGbSipState
 };
 std::string srs_gb_sip_state(SrsGbSipState state);
 
+// For external SIP server mode, where SRS acts only as a media relay server
+//     1. SIP server POST request via HTTP API with stream ID and SSRC
+//     2. SRS create session using ID and SSRC, return a port for receiving media streams (indicated in conf).
+//     3. External streaming service connect to the port, and send RTP stream (with the above SSRC)
+//     4. SRS forward the stream to RTMP stream, named after ID
+//
+// Request:
+//      POST /gb/v1/publish/
+//      {
+//              "id": "...",
+//              "ssrc": "..."
+//      }
+// Response:
+//      {"port":9000, "is_tcp": true}
+class SrsGoApiGbPublish : public ISrsHttpHandler
+{
+private:
+    SrsConfDirective* conf_;
+public:
+    SrsGoApiGbPublish(SrsConfDirective* conf);
+    virtual ~SrsGoApiGbPublish();
+public:
+    virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
+private:
+    virtual srs_error_t do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res);
+    srs_error_t bind_session(std::string stream, uint64_t ssrc);
+};
+
 // The main logic object for GB, the session.
 // Each session contains a SIP object and a media object, that are managed by session. This means session always
 // lives longer than SIP and media, and session will dispose SIP and media when session disposed. In another word,
@@ -191,6 +219,8 @@ public:
 // Interface ISrsTcpHandler
 public:
     virtual srs_error_t on_tcp_client(ISrsListener* listener, srs_netfd_t stfd);
+private:
+    srs_error_t listen_api();
 };
 
 // A GB28181 TCP SIP connection.
@@ -234,6 +264,10 @@ public:
 public:
     // Get the SIP device id.
     std::string device_id();
+    // For use with external SIP signaling server ONLY
+    // When using an external SIP signaling server, device id are not available, so manual configuration is required
+    // This id will be used as the stream name in the RTMP protocol
+    void set_device_id(const std::string& id);
     // Set the cid of all coroutines.
     virtual void set_cid(const SrsContextId& cid);
 private:

trunk/src/core/srs_core_version6.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       6
 #define VERSION_MINOR       0
-#define VERSION_REVISION    143
+#define VERSION_REVISION    144
 
 #endif