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Jetson Inference: WebRTCServer Class Reference

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WebRTCServer Class Reference Utilities Library (jetson-utils) » Networking

WebRTC signalling server for establishing and negotiating connections with peers for bi-directional media streaming. More...

#include <WebRTCServer.h>

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Classes

| | struct | HttpRoute | | | | struct | WebsocketRoute | | |

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Public Types

| | typedef void(* | WebsocketListener) (WebRTCPeer *peer, const char *message, size_t message_size, void *user_data) | | | Function pointer to a callback for handling websocket requests. More...
| | | | typedef SoupServerCallback | HttpListener | | | Function pointer to a callback for handling HTTP requests. More...
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Public Member Functions

| | void | Release () | | | Release a reference to the server instance. More...
| | | | void | AddRoute (const char *path, HttpListener callback, void *user_data=NULL, uint32_t flags=0) | | | Register a function to handle incoming http/www requests at the specified path. More...
| | | | void | AddRoute (const char *path, WebsocketListener callback, void *user_data=NULL, uint32_t flags=0) | | | Register a function to handle incoming Websocket requests at the specified path. More...
| | | | const char * | GetSTUNServer () const | | | Get the STUN server being used. More...
| | | | bool | HasHTTPS () const | | | Return true if the server is using HTTPS. More...
| | | | bool | IsThreaded () const | | | Return true if the server is running in it's own thread. More...
| | | | bool | ProcessRequests (bool blocking=false) | | | Process incoming requests on the server. More...
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Static Public Member Functions

| | static WebRTCServer * | Create (uint16_t port=WEBRTC_DEFAULT_PORT, const char *stun_server=WEBRTC_DEFAULT_STUN_SERVER, const char *ssl_cert=NULL, const char *ssl_key=NULL, bool threaded=true) | | | Create a WebRTC server on this port. More...
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Protected Member Functions

| | | WebRTCServer (uint16_t port, const char *stun_server, const char *ssl_cert_file, const char *ssl_key_file, bool threaded) | | | | | ~WebRTCServer () | | | | bool | init () | | | | HttpRoute * | findHttpRoute (const char *path) const | | | | WebsocketRoute * | findWebsocketRoute (const char *path) const | | | | void | freeRoute (HttpRoute *route) | | | | void | freeRoute (WebsocketRoute *route) | | | | std::string | printRouteInfo (WebsocketRoute *route) const | | |

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Static Protected Member Functions

| | static void * | runThread (void *user_data) | | | | static void | onHttpRequest (SoupServer *soup_server, SoupMessage *message, const char *path, GHashTable *query, SoupClientContext *client_context, void *user_data) | | | | static void | onHttpDefault (SoupServer *soup_server, SoupMessage *message, const char *path, GHashTable *query, SoupClientContext *client_context, void *user_data) | | | | static void | onWebsocketOpened (SoupServer *server, SoupWebsocketConnection *connection, const char *path, SoupClientContext *client_context, void *user_data) | | | | static void | onWebsocketMessage (SoupWebsocketConnection *connection, SoupWebsocketDataType data_type, GBytes *message, void *user_data) | | | | static void | onWebsocketClosed (SoupWebsocketConnection *connection, void *user_data) | | |

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Protected Attributes

| | std::vector< HttpRoute * > | mHttpRoutes | | | | std::vector< WebsocketRoute * > | mWebsocketRoutes | | | | SoupServer * | mSoupServer | | | | std::string | mStunServer | | | | std::string | mSSLCertFile | | | | std::string | mSSLKeyFile | | | | bool | mHasHTTPS | | | | uint16_t | mPort | | | | uint32_t | mRefCount | | | | uint32_t | mPeerCount | | | | Thread * | mThread | | |

Detailed Description

WebRTC signalling server for establishing and negotiating connections with peers for bi-directional media streaming.

Users can bind routing functions that configure their pipelines dynamically to handle incoming/outgoing streams.

It uses websockets to handle the communication with clients for SDP offers, ICE messages, ect.

See alsohttps://doc-kurento.readthedocs.io/en/stable/_static/client-javadoc/org/kurento/client/WebRtcEndpoint.html

By default it also serves simple HTML for viewing video streams in browsers, but for full hosting you'll want to run this alongside an actual webserver.

multi-stream :: multi-client :: full-duplex

Member Typedef Documentation

◆HttpListener

| typedef SoupServerCallback WebRTCServer::HttpListener |

Function pointer to a callback for handling HTTP requests.

See alsohttps://developer-old.gnome.org/libsoup/stable/SoupServer.html#SoupServerCallback

◆WebsocketListener

| typedef void(* WebRTCServer::WebsocketListener) (WebRTCPeer *peer, const char *message, size_t message_size, void *user_data) |

Function pointer to a callback for handling websocket requests.

See alsoAddConnectionListener

Constructor & Destructor Documentation

◆WebRTCServer()

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| WebRTCServer::WebRTCServer | ( | uint16_t | port, | | | | const char * | stun_server, | | | | const char * | ssl_cert_file, | | | | const char * | ssl_key_file, | | | | bool | threaded | | | ) | | |

| protected |

◆~WebRTCServer()

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| WebRTCServer::~WebRTCServer | ( | | ) | |

| protected |

Member Function Documentation

◆AddRoute() [1/2]

| void WebRTCServer::AddRoute | ( | const char * | path, | | | | HttpListener | callback, | | | | void * | user_data = NULL, | | | | uint32_t | flags = 0 | | | ) | | |

Register a function to handle incoming http/www requests at the specified path.

These are for serving webpages, and should be on different paths than the websockets. If the path is "/" the callback will handle all routes. If the route for this path already exists, it will replace the existing route. To remove a route, set the callback to NULL.

◆AddRoute() [2/2]

| void WebRTCServer::AddRoute | ( | const char * | path, | | | | WebsocketListener | callback, | | | | void * | user_data = NULL, | | | | uint32_t | flags = 0 | | | ) | | |

Register a function to handle incoming Websocket requests at the specified path.

This path often represents a media stream that can be sent/recieved, or a function for dynamically configuring streams when new clients connect.

If the path is "/" the callback will handle all routes. If the route for this path already exists, it will replace the existing route. To remove a route, set the callback to NULL.

The optional flags are WebRTCFlags OR'd together and are used for describing the stream capabilities for purposes of discovery.

See alsoWebRTCFlags

◆Create()

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| static WebRTCServer* WebRTCServer::Create | ( | uint16_t | port = WEBRTC_DEFAULT_PORT, | | | | const char * | stun_server = WEBRTC_DEFAULT_STUN_SERVER, | | | | const char * | ssl_cert = NULL, | | | | const char * | ssl_key = NULL, | | | | bool | threaded = true | | | ) | | |

| static |

Create a WebRTC server on this port.

If this port is already in use, the existing server instance will be returned.

◆findHttpRoute()

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| HttpRoute* WebRTCServer::findHttpRoute | ( | const char * | path | ) | const |

| protected |

◆findWebsocketRoute()

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| WebsocketRoute* WebRTCServer::findWebsocketRoute | ( | const char * | path | ) | const |

| protected |

◆freeRoute() [1/2]

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| void WebRTCServer::freeRoute | ( | HttpRoute * | route | ) | |

| protected |

◆freeRoute() [2/2]

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| void WebRTCServer::freeRoute | ( | WebsocketRoute * | route | ) | |

| protected |

◆GetSTUNServer()

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| const char* WebRTCServer::GetSTUNServer | ( | | ) | const |

| inline |

Get the STUN server being used.

STUN servers are used during ICE/NAT and allow a local device to determine its public IP address.

See alsoWEBRTC_DEFAULT_STUN_SERVER

◆HasHTTPS()

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| bool WebRTCServer::HasHTTPS | ( | | ) | const |

| inline |

Return true if the server is using HTTPS.

For HTTPS to be enabled, SSL cert/key files must have been provided to WebRTCServer::Create().

◆init()

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| bool WebRTCServer::init | ( | | ) | |

| protected |

◆IsThreaded()

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| bool WebRTCServer::IsThreaded | ( | | ) | const |

| inline |

Return true if the server is running in it's own thread.

Otherwise, ProcessRequests() must be called periodically.

◆onHttpDefault()

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| static void WebRTCServer::onHttpDefault | ( | SoupServer * | soup_server, | | | | SoupMessage * | message, | | | | const char * | path, | | | | GHashTable * | query, | | | | SoupClientContext * | client_context, | | | | void * | user_data | | | ) | | |

| staticprotected |

◆onHttpRequest()

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| static void WebRTCServer::onHttpRequest | ( | SoupServer * | soup_server, | | | | SoupMessage * | message, | | | | const char * | path, | | | | GHashTable * | query, | | | | SoupClientContext * | client_context, | | | | void * | user_data | | | ) | | |

| staticprotected |

◆onWebsocketClosed()

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| static void WebRTCServer::onWebsocketClosed | ( | SoupWebsocketConnection * | connection, | | | | void * | user_data | | | ) | | |

| staticprotected |

◆onWebsocketMessage()

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| static void WebRTCServer::onWebsocketMessage | ( | SoupWebsocketConnection * | connection, | | | | SoupWebsocketDataType | data_type, | | | | GBytes * | message, | | | | void * | user_data | | | ) | | |

| staticprotected |

◆onWebsocketOpened()

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| static void WebRTCServer::onWebsocketOpened | ( | SoupServer * | server, | | | | SoupWebsocketConnection * | connection, | | | | const char * | path, | | | | SoupClientContext * | client_context, | | | | void * | user_data | | | ) | | |

| staticprotected |

◆printRouteInfo()

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| std::string WebRTCServer::printRouteInfo | ( | WebsocketRoute * | route | ) | const |

| protected |

◆ProcessRequests()

| bool WebRTCServer::ProcessRequests | ( | bool | blocking = false | ) | |

Process incoming requests on the server.

If set to blocking, the function can wait indefinitely for requests. This should only be called externally if the server was created with threaded=false

◆Release()

| void WebRTCServer::Release | ( | | ) | |

Release a reference to the server instance.

Server will be shut down when the reference count reaches zero.

◆runThread()

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| static void* WebRTCServer::runThread | ( | void * | user_data | ) | |

| staticprotected |

Member Data Documentation

◆mHasHTTPS

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| bool WebRTCServer::mHasHTTPS |

| protected |

◆mHttpRoutes

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| std::vector<HttpRoute*> WebRTCServer::mHttpRoutes |

| protected |

◆mPeerCount

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| uint32_t WebRTCServer::mPeerCount |

| protected |

◆mPort

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| uint16_t WebRTCServer::mPort |

| protected |

◆mRefCount

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| uint32_t WebRTCServer::mRefCount |

| protected |

◆mSoupServer

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| SoupServer* WebRTCServer::mSoupServer |

| protected |

◆mSSLCertFile

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| std::string WebRTCServer::mSSLCertFile |

| protected |

◆mSSLKeyFile

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| std::string WebRTCServer::mSSLKeyFile |

| protected |

◆mStunServer

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| std::string WebRTCServer::mStunServer |

| protected |

◆mThread

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| Thread* WebRTCServer::mThread |

| protected |

◆mWebsocketRoutes

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| std::vector<WebsocketRoute*> WebRTCServer::mWebsocketRoutes |

| protected |

The documentation for this class was generated from the following file: