GStreamer GstWebRTCICE

https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c
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### Get STUN Server URI

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the URI of the configured STUN server for the GstWebRTCICE instance. This function is available in C, JavaScript, and Python.

```C
gchar *
gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice)
```

```APIDOC
Function: gst_webrtc_ice_get_stun_server
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
  Returns:
    gchar* (nullable): URI of the STUN server.
  Since: 1.22
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_stun_server(): {
    // javascript wrapper for 'gst_webrtc_ice_get_stun_server'
}
```

```APIDOC
Function: GstWebRTC.WebRTCICE.prototype.get_stun_server
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  Returns:
    String: URI of the STUN server.
  Since: 1.22
```

```Python
def GstWebRTC.WebRTCICE.get_stun_server (self):
    #python wrapper for 'gst_webrtc_ice_get_stun_server'
```

```APIDOC
Function: GstWebRTC.WebRTCICE.get_stun_server
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  Returns:
    str: URI of the STUN server.
  Since: 1.22
```

--------------------------------

### Get TURN Server URI for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the URI of the configured TURN server for a GstWebRTCICE instance. The function returns a nullable string representing the TURN server URI.

```C
gchar *
gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_turn_server(): {
    // javascript wrapper for 'gst_webrtc_ice_get_turn_server'
}
```

```Python
def GstWebRTC.WebRTCICE.get_turn_server (self):
    #python wrapper for 'gst_webrtc_ice_get_turn_server'
```

```APIDOC
gst_webrtc_ice_get_turn_server (C):
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
  Returns:
    gchar* (nullable) - URI of the TURN server.

GstWebRTC.WebRTCICE.prototype.get_turn_server (JavaScript):
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  Returns:
    String - URI of the TURN server.

GstWebRTC.WebRTCICE.get_turn_server (Python):
  Parameters:
    self: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  Returns:
    str - URI of the TURN server.
```

--------------------------------

### Get STUN Server Address for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the STUN server address configured for the GstWebRTCICE instance. This method has C, JavaScript, and Python implementations.

```APIDOC
Method: get_stun_server
Description: Retrieves the STUN server address.
Parameters:
  ice: GstWebRTCICE - No description available
Returns:
  String - No description available
```

```c
gchar *
get_stun_server (GstWebRTCICE * ice)
```

```javascript
function vfunc_get_stun_server(ice: GstWebRTC.WebRTCICE): {
    // javascript implementation of the 'get_stun_server' virtual method
}
```

```python
def do_get_stun_server (ice):
    #python implementation of the 'get_stun_server' virtual method
```

--------------------------------

### Get TURN Server Address for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the TURN server address configured for the GstWebRTCICE instance. This method has C, JavaScript, and Python implementations.

```APIDOC
Method: get_turn_server
Description: Retrieves the TURN server address.
Parameters:
  ice: GstWebRTCICE - No description available
Returns:
  String - No description available
```

```c
gchar *
get_turn_server (GstWebRTCICE * ice)
```

```javascript
function vfunc_get_turn_server(ice: GstWebRTC.WebRTCICE): {
    // javascript implementation of the 'get_turn_server' virtual method
}
```

```python
def do_get_turn_server (ice):
    #python implementation of the 'get_turn_server' virtual method
```

--------------------------------

### Get Selected ICE Candidate Pair (GstWebRTCICE)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the selected local and remote ICE candidate statistics for a given ICE stream. This method is available in C.

```APIDOC
Method: get_selected_pair
Parameters:
  ice: GstWebRTCICE * - No description available
  stream: GstWebRTCICEStream * - No description available
  local_stats: GstWebRTCICECandidateStats ** - No description available
  remote_stats: GstWebRTCICECandidateStats ** - No description available
Returns:
  gboolean - No description available
```

```C
gboolean
get_selected_pair (GstWebRTCICE * ice,
                   GstWebRTCICEStream * stream,
                   GstWebRTCICECandidateStats ** local_stats,
                   GstWebRTCICECandidateStats ** remote_stats)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Priority

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the priority field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_PRIORITY(c) ((c)->prio)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats URL

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the URL field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_URL(c) ((c)->url)
```

--------------------------------

### Get ICE Controller Status (GstWebRTCICE)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves whether the GstWebRTCICE instance is the controller. This method is available as a virtual function in JavaScript and a 'do' method in Python.

```APIDOC
Method: get_is_controller
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
Returns:
  JavaScript: Number - No description available
  Python: bool - No description available
```

```JavaScript
function vfunc_get_is_controller(ice: GstWebRTC.WebRTCICE): {
    // javascript implementation of the 'get_is_controller' virtual method
}
```

```Python
def do_get_is_controller (ice):
    #python implementation of the 'get_is_controller' virtual method
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Protocol

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the protocol field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_PROTOCOL(c) ((c)->proto)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Type

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the type field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_TYPE(c) ((c)->type)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Relay Protocol

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the relay protocol field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_RELAY_PROTOCOL(c) ((c)->relay_proto)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Username Fragment

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the username fragment field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_USERNAME_FRAGMENT(c) ((c)->ABI.abi.username_fragment)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Related Address

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the related address field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_ADDRESS(c) ((c)->ABI.abi.related_address)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats TCP Type

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the TCP type field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_TCP_TYPE(c) ((c)->ABI.abi.tcp_type)
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Related Port

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the related port field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_PORT(c) ((c)->ABI.abi.related_port)
```

--------------------------------

### Get Selected ICE Candidate Pair

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the currently selected ICE candidate pair for a given WebRTC ICE stream, providing statistics for both local and remote candidates. This function is available in C, JavaScript, and Python.

```C
gboolean
gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
                                  GstWebRTCICEStream * stream,
                                  GstWebRTCICECandidateStats ** local_stats,
                                  GstWebRTCICECandidateStats ** remote_stats)
```

```APIDOC
Function: gst_webrtc_ice_get_selected_pair
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
    stream: GstWebRTCICEStream - The GstWebRTCICEStream instance.
    local_stats: GstWebRTCICECandidateStats** (out, transfer: full) - A pointer to GstWebRTCICECandidateStats for local candidate.
    remote_stats: GstWebRTCICECandidateStats** (out, transfer: full) - A pointer to GstWebRTCICECandidateStats for remote candidate.
  Returns:
    gboolean: FALSE on failure, otherwise local_stats and remote_stats will be set.
  Since: 1.22
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_selected_pair(stream: GstWebRTC.WebRTCICEStream): {
    // javascript wrapper for 'gst_webrtc_ice_get_selected_pair'
}
```

```APIDOC
Function: GstWebRTC.WebRTCICE.prototype.get_selected_pair
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream instance.
  Returns:
    tuple:
      Number: Indicates success (non-zero) or failure (zero).
      local_stats: GstWebRTC.WebRTCICECandidateStats - Statistics for the local candidate.
      remote_stats: GstWebRTC.WebRTCICECandidateStats - Statistics for the remote candidate.
  Since: 1.22
```

```Python
def GstWebRTC.WebRTCICE.get_selected_pair (self, stream):
    #python wrapper for 'gst_webrtc_ice_get_selected_pair'
```

```APIDOC
Function: GstWebRTC.WebRTCICE.get_selected_pair
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream instance.
  Returns:
    tuple:
      bool: Indicates success (True) or failure (False).
      local_stats: GstWebRTC.WebRTCICECandidateStats - Statistics for the local candidate.
      remote_stats: GstWebRTC.WebRTCICECandidateStats - Statistics for the remote candidate.
  Since: 1.22
```

--------------------------------

### C Macro to Get GstWebRTC ICE Candidate Stats Stream ID

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Macro to access the stream ID field of a GstWebRTCICE candidate statistics structure. Available since GStreamer 1.28.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_STREAM_ID(c) ((c)->stream_id)
```

--------------------------------

### Get Selected ICE Candidate Pair for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the currently selected ICE candidate pair for a given WebRTCICE stream, including local and remote candidate statistics. This method is a virtual function with JavaScript and Python implementations.

```APIDOC
Method: get_selected_pair
Description: Retrieves the currently selected ICE candidate pair.
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
  stream: GstWebRTC.WebRTCICEStream - No description available
Returns:
  tuple:
    - Number - No description available
    - local_stats: GstWebRTC.WebRTCICECandidateStats - No description available
    - remote_stats: GstWebRTC.WebRTCICECandidateStats - No description available
```

```javascript
function vfunc_get_selected_pair(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream): {
    // javascript implementation of the 'get_selected_pair' virtual method
}
```

```python
def do_get_selected_pair (ice, stream):
    #python implementation of the 'get_selected_pair' virtual method
```

--------------------------------

### Configure STUN Server for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet demonstrates how to set the STUN server URI for a GstWebRTCICE instance. It includes the C function signature, its virtual method implementations in JavaScript and Python, and parameter details.

```C
set_stun_server (GstWebRTCICE * ice,
                 const gchar * uri)
```

```APIDOC
Parameters:
  ice: GstWebRTCICE instance
  uri: The URI of the STUN server (gchar*)
```

```JavaScript
function vfunc_set_stun_server(ice: GstWebRTC.WebRTCICE, uri: String): {
    // javascript implementation of the 'set_stun_server' virtual method
}
```

```Python
def do_set_stun_server (ice, uri):
    #python implementation of the 'set_stun_server' virtual method
```

--------------------------------

### Configure TURN Server for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet demonstrates how to set the TURN server URI for a GstWebRTCICE instance. It includes the C function signature, its virtual method implementations in JavaScript and Python, and parameter details.

```C
set_turn_server (GstWebRTCICE * ice,
                 const gchar * uri)
```

```APIDOC
Parameters:
  ice: GstWebRTCICE instance
  uri: The URI of the TURN server (gchar*)
```

```JavaScript
function vfunc_set_turn_server(ice: GstWebRTC.WebRTCICE, uri: String): {
    // javascript implementation of the 'set_turn_server' virtual method
}
```

```Python
def do_set_turn_server (ice, uri):
    #python implementation of the 'set_turn_server' virtual method
```

--------------------------------

### Configure HTTP Proxy for GstWebRTCICE TURN Connections

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet provides implementations and API documentation for setting an HTTP proxy to be used when connecting to a TURN server. It covers C, JavaScript, and Python versions of the `set_http_proxy` method, detailing the required URI format.

```c
set_http_proxy (GstWebRTCICE * ice,
                const gchar * uri)
```

```javascript
function vfunc_set_http_proxy(ice: GstWebRTC.WebRTCICE, uri: String): {
    // javascript implementation of the 'set_http_proxy' virtual method
}
```

```python
def do_set_http_proxy (ice, uri):
    #python implementation of the 'set_http_proxy' virtual method
```

```APIDOC
set_http_proxy:
  Description: Set HTTP Proxy to be used when connecting to TURN server.
  Parameters:
    ice: GstWebRTCICE | GstWebRTC.WebRTCICE
    uri: const gchar* | String | str
      Description: URI of the HTTP proxy of the form http://[username:password@]hostname[:port]
  Since: 1.22
```

--------------------------------

### Set GstWebRTCICE as Controller

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Configures a GstWebRTCICE instance to act as the controller. Setting `controller` to TRUE designates the instance as the controller.

```C
gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice,
                                  gboolean controller)
```

```APIDOC
gst_webrtc_ice_set_is_controller (C):
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
    controller: gboolean - TRUE to set as controller.
```

--------------------------------

### Configure STUN Server for GstWebRTC.WebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method allows setting the URI for the STUN (Session Traversal Utilities for NAT) server, which is crucial for establishing direct peer-to-peer connections in WebRTC by assisting with NAT traversal.

```javascript
function GstWebRTC.WebRTCICE.prototype.set_stun_server(uri: String): {
    // javascript wrapper for 'gst_webrtc_ice_set_stun_server'
}
```

```python
def GstWebRTC.WebRTCICE.set_stun_server (self, uri):
    #python wrapper for 'gst_webrtc_ice_set_stun_server'
```

```APIDOC
Method: GstWebRTC.WebRTCICE.set_stun_server
Purpose: Sets the URI of the STUN server.
Parameters:
  ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  uri: String (JavaScript), str (Python) - URI of the STUN server.
Since: 1.22
```

--------------------------------

### GStreamer WebRTC Library API Components Overview

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/index

This section provides an overview of the main components within the GStreamer WebRTC Library API, listing various RTC objects and their brief descriptions. It serves as an index to more detailed documentation for each component.

```APIDOC
GstWebRTC Enumerations
GstWebRTCDataChannel – RTCDataChannel object
GstWebRTCDTLSTransport – RTCDtlsTransport object
GstWebRTCICE – Base class WebRTC ICE handling
GstWebRTCICEStream – GstWebRTCICEStream object
GstWebRTCICETransport – RTCIceTransport object
GstWebRTCRTPReceiver – RTCRtpReceiver object
GstWebRTCRTPSender – RTCRtpSender object
GstWebRTCRTPTransceiver – RTCRtpTransceiver object
GstWebRTCSessionDescription – RTCSessionDescription object
webrtc
```

--------------------------------

### Set Local Credentials for GstWebRTCICE Stream

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet provides implementations for setting local credentials (ufrag and pwd) for a specific GstWebRTCICEStream. It includes C and JavaScript versions, detailing the parameters and return types.

```c
gboolean
set_local_credentials (GstWebRTCICE * ice,
                       GstWebRTCICEStream * stream,
                       const gchar * ufrag,
                       const gchar * pwd)
```

```javascript
function vfunc_set_local_credentials(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream, ufrag: String, pwd: String): {
    // javascript implementation of the 'set_local_credentials' virtual method
}
```

```APIDOC
set_local_credentials:
  Parameters:
    ice: GstWebRTCICE | GstWebRTC.WebRTCICE
    stream: GstWebRTCICEStream | GstWebRTC.WebRTCICEStream
    ufrag: const gchar* | String
    pwd: const gchar* | String
  Returns: gboolean | Number
```

--------------------------------

### Gather WebRTC ICE Candidates

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method initiates the gathering of ICE candidates for a given ICE instance and stream. It has implementations in C, JavaScript, and Python.

```C
gboolean
gather_candidates (GstWebRTCICE * ice,
                   GstWebRTCICEStream * stream)
```

```JavaScript
function vfunc_gather_candidates(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream): {
    // javascript implementation of the 'gather_candidates' virtual method
}
```

```Python
def do_gather_candidates (ice, stream):
    #python implementation of the 'gather_candidates' virtual method
```

```APIDOC
Method: gather_candidates
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
  stream: GstWebRTC.WebRTCICEStream - No description available
Returns:
  C: gboolean - No description available
  JavaScript: Number - No description available
  Python: bool - No description available
```

--------------------------------

### Retrieve HTTP Proxy for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the URI of the HTTP proxy configured for connecting to a TURN server. The URI follows the format `http://[username:password@]hostname[:port][?alpn=]`. This method is available in C, JavaScript, and Python.

```C
gchar *
gst_webrtc_ice_get_http_proxy (GstWebRTCICE * ice)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_http_proxy(): {
    // javascript wrapper for 'gst_webrtc_ice_get_http_proxy'
}
```

```Python
def GstWebRTC.WebRTCICE.get_http_proxy (self):
    #python wrapper for 'gst_webrtc_ice_get_http_proxy'
```

```APIDOC
gst_webrtc_ice_get_http_proxy (GstWebRTCICE * ice)
  Description: Get HTTP Proxy to be used when connecting to TURN server.
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
  Returns:
    C: gchar * (transfer: full) - URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=]
    JavaScript: String - URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=]
    Python: str - URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=]
  Since: 1.22
```

--------------------------------

### Configure TURN Server for GstWebRTC.WebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method allows setting the URI for the TURN (Traversal Using Relays around NAT) server. A TURN server is used to relay media when direct peer-to-peer connections cannot be established, providing a fallback mechanism for WebRTC.

```c
gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice,
                                const gchar * uri)
```

```javascript
function GstWebRTC.WebRTCICE.prototype.set_turn_server(uri: String): {
    // javascript wrapper for 'gst_webrtc_ice_set_turn_server'
}
```

```python
def GstWebRTC.WebRTCICE.set_turn_server (self, uri):
    #python wrapper for 'gst_webrtc_ice_set_turn_server'
```

```APIDOC
Method: GstWebRTC.WebRTCICE.set_turn_server
Purpose: Sets the URI of the TURN server.
Parameters:
  ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  uri: const gchar* (C, nullable), String (JavaScript), str (Python) - URI of the TURN server.
Since: 1.22
```

--------------------------------

### Add TURN Server to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method adds a TURN server URI to a GstWebRTCICE instance. It is available in C, as a JavaScript virtual function (`vfunc`), and a Python direct implementation (`do`). It returns a boolean indicating success.

```C
gboolean
add_turn_server (GstWebRTCICE * ice,
                 const gchar * uri)
```

```JavaScript
function vfunc_add_turn_server(ice: GstWebRTC.WebRTCICE, uri: String): {
    // javascript implementation of the 'add_turn_server' virtual method
}
```

```Python
def do_add_turn_server (ice, uri):
    #python implementation of the 'add_turn_server' virtual method
```

```APIDOC
add_turn_server / vfunc_add_turn_server / do_add_turn_server:
  Parameters:
    ice: GstWebRTCICE (C), GstWebRTC.WebRTCICE (JavaScript), GstWebRTC.WebRTCICE (Python)
    uri: const gchar * (C), String (JavaScript), str (Python)
  Returns:
    gboolean (C), Number (JavaScript), bool (Python)
```

--------------------------------

### Set STUN Server for GstWebRTC.WebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Configures the URI of the STUN (Session Traversal Utilities for NAT) server to be used by the GstWebRTCICE instance. A STUN server helps in discovering the public IP address and port of the client behind a NAT.

```C
gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice,
                                const gchar * uri)
```

```APIDOC
gst_webrtc_ice_set_stun_server:
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    uri: nullable string - URI of the STUN server.
  Since: 1.22
```

--------------------------------

### Set HTTP Proxy for GstWebRTCICE TURN Server

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Sets the HTTP proxy to be used when connecting to the TURN server for a GstWebRTCICE instance. The `uri` parameter specifies the proxy URI, which can include username, password, hostname, port, and ALPN.

```C
gst_webrtc_ice_set_http_proxy (GstWebRTCICE * ice,
                               const gchar * uri)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.set_http_proxy(uri: String): {
    // javascript wrapper for 'gst_webrtc_ice_set_http_proxy'
}
```

```Python
def GstWebRTC.WebRTCICE.set_http_proxy (self, uri):
    #python wrapper for 'gst_webrtc_ice_set_http_proxy'
```

```APIDOC
gst_webrtc_ice_set_http_proxy (C):
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
    uri: const gchar* (transfer: none) - URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=].

GstWebRTC.WebRTCICE.prototype.set_http_proxy (JavaScript):
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    uri: String - URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=].

GstWebRTC.WebRTCICE.set_http_proxy (Python):
  Parameters:
    self: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    uri: str - URI of the HTTP proxy of the form http://[username:password@]hostname[:port][?alpn=].
```

--------------------------------

### Set Force Relay for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet demonstrates the `set_force_relay` virtual method, allowing control over whether ICE connections are forced to use relay candidates. It includes the Python implementation and API documentation for its parameters.

```python
def do_set_force_relay (ice, force_relay):
    #python implementation of the 'set_force_relay' virtual method
```

```APIDOC
Parameters:
  ice: GstWebRTC.WebRTCICE
  force_relay: bool
```

--------------------------------

### GstWebRTC.WebRTCICE Class Definition (Python)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the GstWebRTC.WebRTCICE class, showing its inheritance hierarchy and its core members for Python bindings.

```APIDOC
GstWebRTC.WebRTCICE:
  Inheritance:
    - GObject.Object
    -   ╰──GObject.InitiallyUnowned
    -       ╰──Gst.Object
    -           ╰──GstWebRTC.WebRTCICE
  Members:
    - name: parent
      type: Gst.Object
      description: No description available
    - name: ice_gathering_state
      type: GstWebRTC.WebRTCICEGatheringState
      description: No description available
    - name: ice_connection_state
      type: GstWebRTC.WebRTCICEConnectionState
      description: No description available
    - name: min_rtp_port
      type: int
      description: No description available
    - name: max_rtp_port
      type: int
      description: No description available
    - name: _gst_reserved
      type: [ object ]
      description: No description available
  Since: 1.22
```

--------------------------------

### Set Force Relay for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Configures the GstWebRTCICE instance to force relaying of media through a TURN server. This method has C and JavaScript implementations.

```APIDOC
Method: set_force_relay
Description: Configures the GstWebRTCICE instance to force relaying.
Parameters:
  ice: GstWebRTCICE - No description available
  force_relay: gboolean - No description available
```

```c
set_force_relay (GstWebRTCICE * ice,
                 gboolean force_relay)
```

```javascript
function vfunc_set_force_relay(ice: GstWebRTC.WebRTCICE, force_relay: Number): {
    // javascript implementation of the 'set_force_relay' virtual method
}
```

--------------------------------

### Add ICE Candidate to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Adds an ICE candidate to a GstWebRTCICE instance, facilitating peer-to-peer connection establishment. This method is available in JavaScript and Python wrappers, and the underlying C implementation.

```javascript
function GstWebRTC.WebRTCICE.prototype.add_candidate(stream: GstWebRTC.WebRTCICEStream, candidate: String, promise: Gst.Promise): {
    // javascript wrapper for 'gst_webrtc_ice_add_candidate'
}
```

```python
def GstWebRTC.WebRTCICE.add_candidate (self, stream, candidate, promise):
    #python wrapper for 'gst_webrtc_ice_add_candidate'
```

```APIDOC
GstWebRTC.WebRTCICE.prototype.add_candidate (stream: GstWebRTC.WebRTCICEStream, candidate: String, promise: Gst.Promise)
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream.
    candidate: String - The ICE candidate.
    promise: Gst.Promise - A Gst.Promise for task notifications (Since: 1.24).
  Since: 1.22

GstWebRTC.WebRTCICE.add_candidate (self, stream, candidate, promise)
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream.
    candidate: str - The ICE candidate.
    promise: Gst.Promise - A Gst.Promise for task notifications (Since: 1.24).
  Since: 1.22
```

--------------------------------

### GstWebRTCICE Class Definition (C)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the GstWebRTCICE class, showing its inheritance hierarchy and its core members in C.

```APIDOC
GstWebRTCICE:
  Inheritance:
    - GObject
    -   ╰──GInitiallyUnowned
    -       ╰──GstObject
    -           ╰──GstWebRTCICE
  Members:
    - name: parent
      type: GstObject
      description: No description available
    - name: ice_gathering_state
      type: GstWebRTCICEGatheringState
      description: No description available
    - name: ice_connection_state
      type: GstWebRTCICEConnectionState
      description: No description available
    - name: min_rtp_port
      type: guint
      description: No description available
    - name: max_rtp_port
      type: guint
      description: No description available
    - name: _gst_reserved
      type: gpointer *
      description: No description available
  Since: 1.22
```

--------------------------------

### Python Callback for GstWebRTC ICE Candidate Discovery

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the Python wrapper for the GstWebRTCICEOnCandidateFunc callback. This function is invoked when a new ICE candidate is discovered, providing the GstWebRTCICE instance, stream ID, candidate string, and user data. Available since GStreamer 1.22.

```Python
def GstWebRTC.WebRTCICEOnCandidateFunc (ice, stream_id, candidate, *user_data):
    #python wrapper for 'GstWebRTCICEOnCandidateFunc'
```

```APIDOC
Parameters:
  ice (GstWebRTC.WebRTCICE): The GstWebRTC.WebRTCICE
  stream_id (int): The stream id
  candidate (str): The discovered candidate
  user_data (variadic): User data that was set by GstWebRTC.WebRTCICE.set_on_ice_candidate
Since: 1.22
```

--------------------------------

### GstWebRTC.WebRTCICE Class Definition (JavaScript)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the GstWebRTC.WebRTCICE class, showing its inheritance hierarchy and its core members for JavaScript bindings.

```APIDOC
GstWebRTC.WebRTCICE:
  Inheritance:
    - GObject.Object
    -   ╰──GObject.InitiallyUnowned
    -       ╰──Gst.Object
    -           ╰──GstWebRTC.WebRTCICE
  Members:
    - name: parent
      type: Gst.Object
      description: No description available
    - name: ice_gathering_state
      type: GstWebRTC.WebRTCICEGatheringState
      description: No description available
    - name: ice_connection_state
      type: GstWebRTC.WebRTCICEConnectionState
      description: No description available
    - name: min_rtp_port
      type: Number
      description: No description available
    - name: max_rtp_port
      type: Number
      description: No description available
    - name: _gst_reserved
      type: [ Object ]
      description: No description available
  Since: 1.22
```

--------------------------------

### Set Local Credentials for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method sets the local ICE credentials (ufrag and password) for a given GstWebRTCICE instance and stream. It is the Python implementation of the 'set_local_credentials' virtual method.

```Python
def do_set_local_credentials (ice, stream, ufrag, pwd):
    #python implementation of the 'set_local_credentials' virtual method
```

```APIDOC
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
  stream: GstWebRTC.WebRTCICEStream - No description available
  ufrag: str - No description available
  pwd: str - No description available
Returns:
  bool - No description available
```

--------------------------------

### Set Force Relay for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Configures whether to force relaying for a GstWebRTCICE instance. Setting `force_relay` to TRUE enables forced relaying.

```C
gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice,
                                gboolean force_relay)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.set_force_relay(force_relay: Number): {
    // javascript wrapper for 'gst_webrtc_ice_set_force_relay'
}
```

```Python
def GstWebRTC.WebRTCICE.set_force_relay (self, force_relay):
    #python wrapper for 'gst_webrtc_ice_set_force_relay'
```

```APIDOC
gst_webrtc_ice_set_force_relay (C):
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
    force_relay: gboolean - TRUE to enable force relay.

GstWebRTC.WebRTCICE.prototype.set_force_relay (JavaScript):
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    force_relay: Number - TRUE to enable force relay.

GstWebRTC.WebRTCICE.set_force_relay (Python):
  Parameters:
    self: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    force_relay: bool - TRUE to enable force relay.
```

--------------------------------

### Add TURN Server to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Adds a TURN server URI to the GstWebRTCICE instance. This helps in traversing NATs and firewalls for WebRTC connections. The method returns a boolean indicating success or failure.

```c
gboolean
gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice,
                                const gchar * uri)
```

```APIDOC
gst_webrtc_ice_add_turn_server (ice: GstWebRTCICE *, uri: const gchar *)
  Parameters:
    ice: GstWebRTCICE - The GstWebRTC.WebRTCICE instance.
    uri: const gchar * - URI of the TURN server.
  Returns:
    gboolean - FALSE on error, TRUE otherwise.
  Since: 1.22
```

--------------------------------

### GstWebRTCICECandidateStats API Reference

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This section provides a detailed reference for the GstWebRTCICECandidateStats structure, which contains statistics about an ICE candidate. It includes member definitions, their data types, and descriptions for both Glib/C and JavaScript contexts, highlighting properties such as IP address, port, candidate type, and protocol.

```APIDOC
GstWebRTCICECandidateStats Members (Glib/C):
  ipaddr (gchar *): A string containing the address of the candidate. This value may be an IPv4 address, an IPv6 address, or a fully-qualified domain name
  port (guint): The network port number used by the candidate
  stream_id (guint): A string that uniquely identifies the object that is being monitored to produce this set of statistics
  type (const gchar *): The candidate type
  proto (const gchar *): A string specifying the protocol (tcp or udp) used to transmit data on the port
  relay_proto (const gchar *): No description available
  prio (guint): The candidate's priority, corresponding to RTCIceCandidate.priority
  url (gchar *): For local candidates, the url property is the URL of the ICE server from which the candidate was received
  ABI.abi.foundation (gchar *): The foundation of the ICE candidate.
  ABI.abi.related_address (gchar *): The related address (STUN or TURN server) of the candidate. Will be NULL for host candidates.
  ABI.abi.related_port (guint): The related port (STUN or TURN server) of the candidate. Will be 0 for host candidates.
  ABI.abi.username_fragment (gchar *): The ICE username for this candidate.
  ABI.abi.tcp_type (GstWebRTCICETcpCandidateType): The type of TCP candidate. Will be NULL if the candidate is not a TCP candidate. (Since: 1.22)
```

```APIDOC
GstWebRTC.WebRTCICECandidateStats Members (JavaScript):
  ipaddr (String): A string containing the address of the candidate. This value may be an IPv4 address, an IPv6 address, or a fully-qualified domain name
  port (Number): The network port number used by the candidate
  stream_id (Number): A string that uniquely identifies the object that is being monitored to produce this set of statistics
  type (String): The candidate type
  proto (String): A string specifying the protocol (tcp or udp) used to transmit data on the port
  relay_proto (String): No description available
  prio (Number): The candidate's priority, corresponding to RTCIceCandidate.priority
  url (String): For local candidates, the url property is the URL of the ICE server from which the candidate was received (Since: 1.22)
```

--------------------------------

### Set ICE Candidate Callback for GstWebRTC.WebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Configures a callback function to be invoked when an ICE candidate is generated. This allows applications to handle and exchange ICE candidates with the remote peer, crucial for establishing WebRTC connections.

```JavaScript
function GstWebRTC.WebRTCICE.prototype.set_on_ice_candidate(func: GstWebRTC.WebRTCICEOnCandidateFunc, user_data: Object): {
    // javascript wrapper for 'gst_webrtc_ice_set_on_ice_candidate'
}
```

```Python
def GstWebRTC.WebRTCICE.set_on_ice_candidate (self, func, *user_data):
    #python wrapper for 'gst_webrtc_ice_set_on_ice_candidate'
```

```APIDOC
GstWebRTC.WebRTCICE.prototype.set_on_ice_candidate:
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    func: GstWebRTC.WebRTCICEOnCandidateFunc - The callback function to be invoked.
    user_data: Object - User data passed to the callback function.
  Since: 1.22
```

--------------------------------

### C Callback for GstWebRTC ICE Candidate Discovery

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the C callback function signature for handling the discovery of new ICE candidates in GstWebRTCICE. This function is invoked when a new candidate is found, providing the ICE instance, stream ID, and the candidate string. Available since GStreamer 1.22.

```C
(*GstWebRTCICEOnCandidateFunc) (GstWebRTCICE * ice,
                                guint stream_id,
                                const gchar * candidate,
                                gpointer user_data)
```

```APIDOC
Parameters:
  ice: The GstWebRTCICE
  stream_id: The stream id
  candidate: The discovered candidate
  user_data: User data that was set by gst_webrtc_ice_set_on_ice_candidate
Since: 1.22
```

--------------------------------

### Gather ICE Candidates for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Initiates the process of gathering ICE candidates for a specified GstWebRTCICE instance and stream. This is a fundamental step in establishing peer-to-peer connections, allowing the system to discover and collect potential network paths for communication.

```C
gboolean
gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
                                  GstWebRTCICEStream * stream)
```

```APIDOC
gst_webrtc_ice_gather_candidates(ice: GstWebRTCICE, stream: GstWebRTCICEStream)
  Parameters:
    ice: The GstWebRTCICE instance
    stream: The GstWebRTCICEStream
  Returns: gboolean (FALSE on error, TRUE otherwise)
  Since: 1.22
```

--------------------------------

### Set Remote Credentials for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

These methods are used to set the remote ICE credentials (ufrag and password) for a GstWebRTCICE instance and stream. They provide implementations for C, JavaScript, and Python, corresponding to the 'set_remote_credentials' virtual method.

```C
gboolean
set_remote_credentials (GstWebRTCICE * ice,
                        GstWebRTCICEStream * stream,
                        const gchar * ufrag,
                        const gchar * pwd)
```

```JavaScript
function vfunc_set_remote_credentials(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream, ufrag: String, pwd: String): {
    // javascript implementation of the 'set_remote_credentials' virtual method
}
```

```Python
def do_set_remote_credentials (ice, stream, ufrag, pwd):
    #python implementation of the 'set_remote_credentials' virtual method
```

```APIDOC
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
  stream: GstWebRTC.WebRTCICEStream - No description available
  ufrag: String - No description available
  pwd: String - No description available
Returns:
  Number | bool - No description available
```

--------------------------------

### Set ICE Candidate Callback for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This set of methods allows setting a callback function to be invoked when an ICE candidate is generated. It includes implementations for C, JavaScript, and Python, serving as the virtual method for handling ICE candidate events.

```C
set_on_ice_candidate (GstWebRTCICE * ice,
                      GstWebRTCICEOnCandidateFunc func,
                      gpointer user_data,
                      GDestroyNotify notify)
```

```JavaScript
function vfunc_set_on_ice_candidate(ice: GstWebRTC.WebRTCICE, func: GstWebRTC.WebRTCICEOnCandidateFunc, user_data: Object): {
    // javascript implementation of the 'set_on_ice_candidate' virtual method
}
```

```Python
def do_set_on_ice_candidate (ice, func, *user_data):
    #python implementation of the 'set_on_ice_candidate' virtual method
```

```APIDOC
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
  func: GstWebRTC.WebRTCICEOnCandidateFunc - No description available
  user_data: Object | variadic | gpointer - No description available
  notify: GDestroyNotify (C-specific) - No description available
```

--------------------------------

### Gather ICE Candidates for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Initiates the gathering of ICE candidates for a given WebRTCICEStream. This method is available through JavaScript and Python bindings, and its API specification details parameters and return values.

```JavaScript
function GstWebRTC.WebRTCICE.prototype.gather_candidates(stream: GstWebRTC.WebRTCICEStream): {
    // javascript wrapper for 'gst_webrtc_ice_gather_candidates'
}
```

```Python
def GstWebRTC.WebRTCICE.gather_candidates (self, stream):
    #python wrapper for 'gst_webrtc_ice_gather_candidates'
```

```APIDOC
GstWebRTC.WebRTCICE.prototype.gather_candidates(stream: GstWebRTC.WebRTCICEStream)
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream.
  Returns:
    JavaScript: Number - FALSE on error, TRUE otherwise.
    Python: bool - FALSE on error, TRUE otherwise.
  Since: 1.22
```

--------------------------------

### Set Controller Role for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet illustrates the `set_is_controller` method, which configures whether a GstWebRTCICE instance acts as a controller. It includes C, JavaScript, and Python implementations along with API parameter details.

```c
set_is_controller (GstWebRTCICE * ice,
                   gboolean controller)
```

```javascript
function vfunc_set_is_controller(ice: GstWebRTC.WebRTCICE, controller: Number): {
    // javascript implementation of the 'set_is_controller' virtual method
}
```

```python
def do_set_is_controller (ice, controller):
    #python implementation of the 'set_is_controller' virtual method
```

```APIDOC
set_is_controller:
  Parameters:
    ice: GstWebRTCICE | GstWebRTC.WebRTCICE
    controller: gboolean | Number | bool
```

--------------------------------

### Add TURN Server to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Adds a TURN server URI to the GstWebRTCICE instance. This function is crucial for enabling NAT traversal and establishing peer-to-peer connections in complex network environments by providing a relay server.

```JavaScript
function GstWebRTC.WebRTCICE.prototype.add_turn_server(uri: String): {
    // javascript wrapper for 'gst_webrtc_ice_add_turn_server'
}
```

```Python
def GstWebRTC.WebRTCICE.add_turn_server (self, uri):
    #python wrapper for 'gst_webrtc_ice_add_turn_server'
```

```APIDOC
GstWebRTC.WebRTCICE.add_turn_server(uri: String/str)
  Parameters:
    uri: URI of the TURN server
  Returns: Number/bool (FALSE on error, TRUE otherwise)
  Since: 1.22
```

--------------------------------

### JavaScript Callback for GstWebRTC ICE Candidate Discovery

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the JavaScript wrapper for the GstWebRTCICEOnCandidateFunc callback. This function is triggered upon discovery of a new ICE candidate, providing the GstWebRTCICE instance, stream ID, candidate string, and user data. Available since GStreamer 1.22.

```JavaScript
function GstWebRTC.WebRTCICEOnCandidateFunc(ice: GstWebRTC.WebRTCICE, stream_id: Number, candidate: String, user_data: Object): {
    // javascript wrapper for 'GstWebRTCICEOnCandidateFunc'
}
```

```APIDOC
Parameters:
  ice (GstWebRTC.WebRTCICE): The GstWebRTC.WebRTCICE
  stream_id (Number): The stream id
  candidate (String): The discovered candidate
  user_data (Object): User data that was set by GstWebRTC.WebRTCICE.prototype.set_on_ice_candidate
Since: 1.22
```

--------------------------------

### GstWebRTC.WebRTCICEClass Attributes (Python)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Details the attributes of the GstWebRTC.WebRTCICEClass for Python bindings.

```APIDOC
GstWebRTC.WebRTCICEClass:
  Attributes:
    - name: parent_class
      type: Gst.ObjectClass
      description: No description available
    - name: _gst_reserved
      type: [ object ]
      description: No description available
```

--------------------------------

### GstWebRTCICEClass Structure (C)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Describes the structure of the GstWebRTCICEClass, including its parent class and reserved members in C.

```APIDOC
GstWebRTCICEClass:
  Fields:
    - name: parent_class
      type: GstObjectClass
      description: No description available
    - name: _gst_reserved
      type: gpointer *
      description: No description available
```

--------------------------------

### Set GstWebRTC ICE Candidate Callback

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This function registers a callback function to be invoked when an ICE candidate is generated. This allows applications to handle and exchange ICE candidates with the remote peer. It takes the ICE instance, the callback function, user data, and a destroy notification.

```C
gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
                                     GstWebRTCICEOnCandidateFunc func,
                                     gpointer user_data,
                                     GDestroyNotify notify)
```

```APIDOC
GstWebRTC.WebRTCICE.set_on_ice_candidate
  Description: Sets a callback function to be invoked when an ICE candidate is generated.
  Parameters:
    ice: (GstWebRTC.WebRTCICE) The GstWebRTC.WebRTCICE instance.
    func: (GstWebRTCICEOnCandidateFunc) The callback function.
    user_data: (gpointer) User data passed to the callback function.
    notify: (GDestroyNotify) A GDestroyNotify when the candidate is no longer needed.
  Since: 1.22
```

--------------------------------

### Add ICE Candidate to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method adds an ICE candidate to a GstWebRTCICE instance. It is available as a virtual function (`vfunc`) in JavaScript and a direct implementation (`do`) in Python. The method takes the ICE instance, stream, candidate string, and a Gst.Promise.

```JavaScript
function vfunc_add_candidate(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream, candidate: String, promise: Gst.Promise): {
    // javascript implementation of the 'add_candidate' virtual method
}
```

```Python
def do_add_candidate (ice, stream, candidate, promise):
    #python implementation of the 'add_candidate' virtual method
```

```APIDOC
vfunc_add_candidate / do_add_candidate:
  Parameters:
    ice: GstWebRTC.WebRTCICE
    stream: GstWebRTC.WebRTCICEStream
    candidate: String (JavaScript), str (Python)
    promise: Gst.Promise
```

--------------------------------

### Copy GstWebRTCICECandidateStats Instance

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Creates a new copy of an existing GstWebRTCICECandidateStats instance. This is useful for safely manipulating statistics without affecting the original object.

```C
GstWebRTCICECandidateStats *
gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats * stats)

Parameters:
  stats: GstWebRTCICECandidateStats * - The GstWebRTCICE instance to copy.
Returns:
  GstWebRTCICECandidateStats * (transfer: full) - A copy of stats.
Since: 1.22
```

```JavaScript
function GstWebRTC.WebRTCICECandidateStats.prototype.copy(): {
    // javascript wrapper for 'gst_webrtc_ice_candidate_stats_copy'
}

Parameters:
  stats: GstWebRTC.WebRTCICECandidateStats - The GstWebRTC.WebRTCICE instance to copy.
Returns:
  GstWebRTC.WebRTCICECandidateStats - A copy of stats.
Since: 1.22
```

```Python
def GstWebRTC.WebRTCICECandidateStats.copy (self):
    #python wrapper for 'gst_webrtc_ice_candidate_stats_copy'

Parameters:
  stats: GstWebRTC.WebRTCICECandidateStats - The GstWebRTC.WebRTCICE instance to copy.
Returns:
  GstWebRTC.WebRTCICECandidateStats - A copy of stats.
Since: 1.22
```

--------------------------------

### Set GstWebRTC ICE Local Credentials

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This function sets the local ICE username fragment (ufrag) and password (pwd) for a specific ICE stream. These credentials are used during the ICE negotiation process to authenticate the local endpoint. It returns FALSE on error and TRUE otherwise.

```C
gboolean
gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
                                      GstWebRTCICEStream * stream,
                                      const gchar * ufrag,
                                      const gchar * pwd)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.set_local_credentials(stream: GstWebRTC.WebRTCICEStream, ufrag: String, pwd: String): {
    // javascript wrapper for 'gst_webrtc_ice_set_local_credentials'
}
```

```Python
def GstWebRTC.WebRTCICE.set_local_credentials (self, stream, ufrag, pwd):
    #python wrapper for 'gst_webrtc_ice_set_local_credentials'
```

```APIDOC
GstWebRTC.WebRTCICE.set_local_credentials
  Description: Sets the local ICE username fragment and password.
  Parameters:
    ice: (GstWebRTC.WebRTCICE) The GstWebRTC.WebRTCICE instance.
    stream: (GstWebRTC.WebRTCICEStream) The GstWebRTC.WebRTCICEStream instance.
    ufrag: (gchar*/String/str) ICE username fragment.
    pwd: (gchar*/String/str) ICE password.
  Returns:
    (gboolean/Number/bool) FALSE on error, TRUE otherwise.
  Since: 1.22
```

--------------------------------

### GstWebRTCICECandidateStats Access Macros

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Provides convenience macros for directly accessing specific member fields of a GstWebRTCICECandidateStats structure, such as IP address, foundation, and port.

```C
#define GST_WEBRTC_ICE_CANDIDATE_STATS_ADDRESS(c) ((c)->ipaddr)
#define GST_WEBRTC_ICE_CANDIDATE_STATS_FOUNDATION(c) ((c)->ABI.abi.foundation)
#define GST_WEBRTC_ICE_CANDIDATE_STATS_PORT(c) ((c)->port)
```

--------------------------------

### Retrieve HTTP Proxy for WebRTC ICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method retrieves the HTTP proxy URI to be used when connecting to a TURN server. It is available in C, JavaScript, and Python, and was introduced in version 1.22.

```C
gchar *
get_http_proxy (GstWebRTCICE * ice)
```

```JavaScript
function vfunc_get_http_proxy(ice: GstWebRTC.WebRTCICE): {
    // javascript implementation of the 'get_http_proxy' virtual method
}
```

```Python
def do_get_http_proxy (ice):
    #python implementation of the 'get_http_proxy' virtual method
```

```APIDOC
Method: get_http_proxy
Parameters:
  ice: GstWebRTC.WebRTCICE - a GstWebRTCICE
Returns:
  C: gchar * (transfer: full) - URI of the HTTP proxy of the form http://[username:password@]hostname[:port]
  JavaScript: String - URI of the HTTP proxy of the form http://[username:password@]hostname[:port]
  Python: str - URI of the HTTP proxy of the form http://[username:password@]hostname[:port]
Since: 1.22
```

--------------------------------

### Find Transport for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method finds a transport for a given GstWebRTCICE instance, stream, and component. It is provided in C.

```C
GstWebRTCICETransport *
find_transport (GstWebRTCICE * ice,
                GstWebRTCICEStream * stream,
                GstWebRTCICEComponent component)
```

```APIDOC
find_transport:
  Parameters:
    ice: GstWebRTCICE
    stream: GstWebRTCICEStream
    component: GstWebRTCICEComponent
  Returns:
    GstWebRTCICETransport
```

--------------------------------

### Add ICE Candidate to GstWebRTCICE (C)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Adds an ICE candidate to a GstWebRTCICE instance. This method is used to provide ICE candidates received from a remote peer to the local ICE agent.

```APIDOC
gst_webrtc_ice_add_candidate:
  Signature: gst_webrtc_ice_add_candidate (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * candidate, GstPromise * promise)
  Parameters:
    - name: ice
      type: GstWebRTCICE
      description: The GstWebRTCICE instance.
    - name: stream
      type: GstWebRTCICEStream
      description: The GstWebRTCICEStream associated with the candidate.
    - name: candidate
      type: const gchar *
      description: The ICE candidate string.
    - name: promise
      type: GstPromise (nullable)
      description: A GstPromise for task notifications (Since: 1.24).
  Since: 1.22
```

--------------------------------

### Set Remote ICE Credentials for GstWebRTC.WebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Sets the remote ICE username fragment (ufrag) and password (pwd) for a specific ICE stream. These credentials are used during the ICE negotiation process to authenticate the remote peer.

```C
gboolean
gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
                                       GstWebRTCICEStream * stream,
                                       const gchar * ufrag,
                                       const gchar * pwd)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.set_remote_credentials(stream: GstWebRTC.WebRTCICEStream, ufrag: String, pwd: String): {
    // javascript wrapper for 'gst_webrtc_ice_set_remote_credentials'
}
```

```Python
def GstWebRTC.WebRTCICE.set_remote_credentials (self, stream, ufrag, pwd):
    #python wrapper for 'gst_webrtc_ice_set_remote_credentials'
```

```APIDOC
gst_webrtc_ice_set_remote_credentials / GstWebRTC.WebRTCICE.set_remote_credentials:
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream.
    ufrag: string - ICE username fragment.
    pwd: string - ICE password.
  Returns:
    boolean - FALSE on error, TRUE otherwise.
  Since: 1.22
```

--------------------------------

### Add Stream to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method adds a new stream to a GstWebRTCICE instance, identified by a session ID. It has C, JavaScript virtual function (`vfunc`), and Python direct implementation (`do`) variants. It returns a GstWebRTCICEStream.

```C
GstWebRTCICEStream *
add_stream (GstWebRTCICE * ice,
            guint session_id)
```

```JavaScript
function vfunc_add_stream(ice: GstWebRTC.WebRTCICE, session_id: Number): {
    // javascript implementation of the 'add_stream' virtual method
}
```

```Python
def do_add_stream (ice, session_id):
    #python implementation of the 'add_stream' virtual method
```

```APIDOC
add_stream / vfunc_add_stream / do_add_stream:
  Parameters:
    ice: GstWebRTCICE (C), GstWebRTC.WebRTCICE (JavaScript), GstWebRTC.WebRTCICE (Python)
    session_id: guint (C), Number (JavaScript), int (Python)
  Returns:
    GstWebRTCICEStream
```

--------------------------------

### GstWebRTC.WebRTCICEClass Attributes (JavaScript)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Details the attributes of the GstWebRTC.WebRTCICEClass for JavaScript bindings.

```APIDOC
GstWebRTC.WebRTCICEClass:
  Attributes:
    - name: parent_class
      type: Gst.ObjectClass
      description: No description available
    - name: _gst_reserved
      type: [ Object ]
      description: No description available
```

--------------------------------

### Retrieve Local ICE Candidates for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This function retrieves a list of local ICE candidates associated with a specific GstWebRTCICE instance and stream. It is essential for understanding the local network interfaces and addresses used for WebRTC connections. The function requires the ICE agent and the specific stream for which candidates are to be retrieved.

```C
GstWebRTCICECandidateStats **
gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
                                     GstWebRTCICEStream * stream)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_local_candidates(stream: GstWebRTC.WebRTCICEStream): {
    // javascript wrapper for 'gst_webrtc_ice_get_local_candidates'
}
```

```Python
def GstWebRTC.WebRTCICE.get_local_candidates (self, stream):
    #python wrapper for 'gst_webrtc_ice_get_local_candidates'
```

```APIDOC
Class: GstWebRTCICE
  Method: get_local_candidates
    Description: Retrieves a list of local ICE candidates.
    Parameters:
      ice: GstWebRTCICE (C) / GstWebRTC.WebRTCICE (JS/Python) - The GstWebRTCICE instance.
      stream: GstWebRTCICEStream (C) / GstWebRTC.WebRTCICEStream (JS/Python) - The GstWebRTCICEStream instance.
    Returns:
      GstWebRTCICECandidateStats[] (C) / GstWebRTC.WebRTCICECandidateStats[] (JS/Python) - List of local candidates.
    Since: 1.22
```

--------------------------------

### GstWebRTCICE Virtual Method: add_candidate

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

A virtual method to add an ICE candidate to a specific ICE stream. This method is typically implemented by subclasses to handle candidate addition logic.

```APIDOC
Virtual Method: add_candidate
  Signature: add_candidate (GstWebRTCICE * ice, GstWebRTCICEStream * stream, const gchar * candidate, GstPromise * promise)
  Parameters:
    ice: GstWebRTCICE * - No description available.
    stream: GstWebRTCICEStream * - No description available.
    candidate: const gchar * - No description available.
    promise: GstPromise * - No description available.
```

```C
add_candidate (GstWebRTCICE * ice,
               GstWebRTCICEStream * stream,
               const gchar * candidate,
               GstPromise * promise)
```

--------------------------------

### Configure Type of Service (ToS) for GstWebRTCICE Stream

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This snippet shows how to set the Type of Service (ToS) value for a specific GstWebRTCICEStream within a GstWebRTCICE instance. It provides the C function signature, its virtual method implementations in JavaScript and Python, and parameter details.

```C
set_tos (GstWebRTCICE * ice,
         GstWebRTCICEStream * stream,
         guint tos)
```

```APIDOC
Parameters:
  ice: GstWebRTCICE instance
  stream: The GstWebRTCICEStream to configure
  tos: The Type of Service (ToS) value (guint)
```

```JavaScript
function vfunc_set_tos(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream, tos: Number): {
    // javascript implementation of the 'set_tos' virtual method
}
```

```Python
def do_set_tos (ice, stream, tos):
    #python implementation of the 'set_tos' virtual method
```

--------------------------------

### GstWebRTCICECandidateStats Members Definition

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the properties and their types for the GstWebRTCICECandidateStats structure, including network address, port, stream ID, candidate type, protocol, relay protocol, priority, and URL.

```APIDOC
GstWebRTCICECandidateStats Members:
  ipaddr (str): A string containing the address of the candidate. This value may be an IPv4 address, an IPv6 address, or a fully-qualified domain name.
  port (int): The network port number used by the candidate.
  stream_id (int): A string that uniquely identifies the object that is being monitored to produce this set of statistics.
  type (str): The candidate type.
  proto (str): A string specifying the protocol (tcp or udp) used to transmit data on the port.
  relay_proto (str): No description available.
  prio (int): The candidate's priority, corresponding to RTCIceCandidate.priority.
  url (str): For local candidates, the url property is the URL of the ICE server from which the candidate was received.
```

--------------------------------

### Retrieve Remote ICE Candidates for GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This function retrieves a list of remote ICE candidates associated with a specific GstWebRTCICE instance and stream. These candidates represent the network addresses and protocols offered by the remote peer for establishing a WebRTC connection. The function requires the ICE agent and the specific stream for which remote candidates are to be retrieved.

```C
GstWebRTCICECandidateStats **
gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
                                      GstWebRTCICEStream * stream)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_remote_candidates(stream: GstWebRTC.WebRTCICEStream): {
    // javascript wrapper for 'gst_webrtc_ice_get_remote_candidates'
}
```

```Python
def GstWebRTC.WebRTCICE.get_remote_candidates (self, stream):
    #python wrapper for 'gst_webrtc_ice_get_remote_candidates'
```

```APIDOC
Class: GstWebRTCICE
  Method: get_remote_candidates
    Description: Retrieves a list of remote ICE candidates.
    Parameters:
      ice: GstWebRTCICE (C) / GstWebRTC.WebRTCICE (JS/Python) - The GstWebRTCICE instance.
      stream: GstWebRTCICEStream (C) / GstWebRTC.WebRTCICEStream (JS/Python) - The GstWebRTCICEStream instance.
    Returns:
      GstWebRTCICECandidateStats[] (C) / GstWebRTC.WebRTCICECandidateStats[] (JS/Python) - List of remote candidates.
    Since: 1.22
```

--------------------------------

### Retrieve Local ICE Candidates (GstWebRTCICE)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the local ICE candidate statistics for a given ICE stream. This method is available in C, as a virtual function in JavaScript, and as a 'do' method in Python.

```APIDOC
Method: get_local_candidates
Parameters:
  ice: GstWebRTC.WebRTCICE / GstWebRTCICE * - No description available
  stream: GstWebRTC.WebRTCICEStream / GstWebRTCICEStream * - No description available
Returns:
  GstWebRTCICECandidateStats / GstWebRTCICECandidateStats ** - No description available
```

```C
GstWebRTCICECandidateStats **
get_local_candidates (GstWebRTCICE * ice,
                      GstWebRTCICEStream * stream)
```

```JavaScript
function vfunc_get_local_candidates(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream): {
    // javascript implementation of the 'get_local_candidates' virtual method
}
```

```Python
def do_get_local_candidates (ice, stream):
    #python implementation of the 'get_local_candidates' virtual method
```

--------------------------------

### Configure Type of Service (ToS) for GstWebRTCICE Stream

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method sets the Type of Service (ToS) value for a specific GstWebRTCICEStream, enabling network traffic prioritization for the stream's packets. This can be used to optimize quality of service.

```c
gst_webrtc_ice_set_tos (GstWebRTCICE * ice,
                        GstWebRTCICEStream * stream,
                        guint tos)
```

```javascript
function GstWebRTC.WebRTCICE.prototype.set_tos(stream: GstWebRTC.WebRTCICEStream, tos: Number): {
    // javascript wrapper for 'gst_webrtc_ice_set_tos'
}
```

```python
def GstWebRTC.WebRTCICE.set_tos (self, stream, tos):
    #python wrapper for 'gst_webrtc_ice_set_tos'
```

```APIDOC
Method: GstWebRTC.WebRTCICE.set_tos
Purpose: Sets the Type of Service (ToS) for an ICE stream.
Parameters:
  ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
  stream: GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream.
  tos: guint (C), Number (JavaScript), int (Python) - ToS value to be set.
Since: 1.22
```

--------------------------------

### Find WebRTC ICE Transport Method

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method is used to find a WebRTC ICE transport for a given ICE stream and component. It is a virtual method with implementations shown for JavaScript and Python.

```JavaScript
function vfunc_find_transport(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream, component: GstWebRTC.WebRTCICEComponent): {
    // javascript implementation of the 'find_transport' virtual method
}
```

```Python
def do_find_transport (ice, stream, component):
    #python implementation of the 'find_transport' virtual method
```

```APIDOC
Method: find_transport
Parameters:
  ice: GstWebRTC.WebRTCICE - No description available
  stream: GstWebRTC.WebRTCICEStream - No description available
  component: GstWebRTC.WebRTCICEComponent - No description available
Returns:
  GstWebRTC.WebRTCICETransport - No description available
```

--------------------------------

### GstWebRTCICE Signal: add-local-ip-address

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Emits the 'add-local-ip-address' signal on a GstWebRTCICE instance to manually add a local IP address for ICE candidate gathering. If no addresses are supplied, they are discovered automatically. Calling this signal stops automatic ICE gathering. The signal returns whether the address could be added.

```APIDOC
Signal: add-local-ip-address
  Description: Add a local IP address to use for ICE candidate gathering. If none are supplied, they will be discovered automatically. Calling this signal stops automatic ICE gathering.
  Parameters:
    self: GstWebRTCICE * (C), GstWebRTC.WebRTCICE (JS/Python) - Instance of GstWebRTCICE.
    address: gchar * (C), String (JS), str (Python) - The local IP address.
    user_data: gpointer (C), Object (JS), variadic (Python) - User data.
  Returns: gboolean (C), Number (JS), bool (Python) - Whether the address could be added.
  Flags: Run Last, Action
```

```C
g_signal_emit_by_name (self, "add-local-ip-address", address, user_data, &ret);
```

```JavaScript
let ret = self.emit ("add-local-ip-address", address, user_data);
```

```Python
ret = self.emit ("add-local-ip-address", address, user_data)
```

--------------------------------

### Check GstWebRTCICE Controller Status

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Determines if the GstWebRTCICE instance is currently set as the controller. This method is accessible via C, JavaScript, and Python bindings, returning a boolean value.

```C
gboolean
gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.get_is_controller(): {
    // javascript wrapper for 'gst_webrtc_ice_get_is_controller'
}
```

```Python
def GstWebRTC.WebRTCICE.get_is_controller (self):
    #python wrapper for 'gst_webrtc_ice_get_is_controller'
```

```APIDOC
gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice)
  Parameters:
    ice: GstWebRTCICE - The GstWebRTCICE instance.
  Returns:
    C: gboolean - TRUE if set as controller, FALSE otherwise.
    JavaScript: Number - TRUE if set as controller, FALSE otherwise.
    Python: bool - TRUE if set as controller, FALSE otherwise.
  Since: 1.22
```

--------------------------------

### Set GstWebRTC ICE Controller Role

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This function sets whether the GstWebRTC ICE agent should act as a controller. The controller role is crucial in the ICE negotiation process. It takes a boolean value to determine the role.

```JavaScript
function GstWebRTC.WebRTCICE.prototype.set_is_controller(controller: Number): {
    // javascript wrapper for 'gst_webrtc_ice_set_is_controller'
}
```

```Python
def GstWebRTC.WebRTCICE.set_is_controller (self, controller):
    #python wrapper for 'gst_webrtc_ice_set_is_controller'
```

```APIDOC
GstWebRTC.WebRTCICE.set_is_controller
  Description: Sets whether the ICE agent is a controller.
  Parameters:
    controller: (Number/bool) TRUE to set as controller.
  Since: 1.22
```

--------------------------------

### Retrieve Remote ICE Candidates (GstWebRTCICE)

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Retrieves the remote ICE candidate statistics for a given ICE stream. This method is available in C, as a virtual function in JavaScript, and as a 'do' method in Python.

```APIDOC
Method: get_remote_candidates
Parameters:
  ice: GstWebRTC.WebRTCICE / GstWebRTCICE * - No description available
  stream: GstWebRTC.WebRTCICEStream / GstWebRTCICEStream * - No description available
Returns:
  GstWebRTCICECandidateStats / GstWebRTCICECandidateStats ** - No description available
```

```C
GstWebRTCICECandidateStats **
get_remote_candidates (GstWebRTCICE * ice,
                       GstWebRTCICEStream * stream)
```

```JavaScript
function vfunc_get_remote_candidates(ice: GstWebRTC.WebRTCICE, stream: GstWebRTC.WebRTCICEStream): {
    // javascript implementation of the 'get_remote_candidates' virtual method
}
```

```Python
def do_get_remote_candidates (ice, stream):
    #python implementation of the 'get_remote_candidates' virtual method
```

--------------------------------

### Add Stream to GstWebRTCICE

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Adds a new stream to a GstWebRTCICE instance, identified by a session ID. This is crucial for managing multiple media streams within a WebRTC connection. Available in C, JavaScript, and Python.

```c
GstWebRTCICEStream *
gst_webrtc_ice_add_stream (GstWebRTCICE * ice,
                           guint session_id)
```

```javascript
function GstWebRTC.WebRTCICE.prototype.add_stream(session_id: Number): {
    // javascript wrapper for 'gst_webrtc_ice_add_stream'
}
```

```python
def GstWebRTC.WebRTCICE.add_stream (self, session_id):
    #python wrapper for 'gst_webrtc_ice_add_stream'
```

```APIDOC
gst_webrtc_ice_add_stream (ice: GstWebRTCICE *, session_id: guint)
  Parameters:
    ice: GstWebRTCICE - The GstWebRTC.WebRTCICE instance.
    session_id: guint - The session id.
  Returns:
    GstWebRTCICEStream - The GstWebRTCICEStream, or NULL.
  Since: 1.22

GstWebRTC.WebRTCICE.prototype.add_stream (session_id: Number)
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    session_id: Number - The session id.
  Returns:
    GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream, or null.
  Since: 1.22

GstWebRTC.WebRTCICE.add_stream (self, session_id)
  Parameters:
    ice: GstWebRTC.WebRTCICE - The GstWebRTC.WebRTCICE instance.
    session_id: int - The session id.
  Returns:
    GstWebRTC.WebRTCICEStream - The GstWebRTC.WebRTCICEStream, or None.
  Since: 1.22
```

--------------------------------

### GstWebRTCICE Property: min-rtp-port

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the minimum port number for the local RTP port range. This value must be less than or equal to 'max-rtp-port'.

```APIDOC
Property: min-rtp-port
  Type (C): guint
  Type (JS): Number
  Type (Python): int
  Description: Minimum port for local rtp port range. min-rtp-port must be <= max-rtp-port
  Flags: Read, Write, Construct
  Since: 1.20
```

```C
"min-rtp-port" guint
```

```JavaScript
"min-rtp-port" Number
```

```Python
"self.props.min_rtp_port" int
```

--------------------------------

### Free GstWebRTCICECandidateStats Instance

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Releases the memory allocated for a GstWebRTCICECandidateStats instance. This function should be called when the statistics object is no longer needed to prevent memory leaks.

```C
gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats)

Helper function to free GstWebRTCICECandidateStats

Parameters:
  stats: GstWebRTCICECandidateStats * - The GstWebRTCICECandidateStats to be free'd.
Since: 1.22
```

```JavaScript
function GstWebRTC.WebRTCICECandidateStats.prototype.free(): {
    // javascript wrapper for 'gst_webrtc_ice_candidate_stats_free'
}

Helper function to free GstWebRTC.WebRTCICECandidateStats

Parameters:
  stats: GstWebRTC.WebRTCICECandidateStats - The GstWebRTC.WebRTCICECandidateStats to be free'd.
Since: 1.22
```

```Python
def GstWebRTC.WebRTCICECandidateStats.free (self):
    #python wrapper for 'gst_webrtc_ice_candidate_stats_free'

Helper function to free GstWebRTC.WebRTCICECandidateStats

Parameters:
  stats: GstWebRTC.WebRTCICECandidateStats - The GstWebRTC.WebRTCICECandidateStats to be free'd.
Since: 1.22
```

--------------------------------

### Check if WebRTC ICE is Controller

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

This method checks if the GstWebRTCICE instance is acting as the controller.

```C
gboolean
get_is_controller (GstWebRTCICE * ice)
```

```APIDOC
Method: get_is_controller
Parameters:
  ice: GstWebRTCICE - No description available
Returns:
  gboolean - No description available
```

--------------------------------

### Find GstWebRTCICETransport for Stream Component

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Locates an existing GstWebRTCICETransport for a given ICE stream and component. This method is used to retrieve the transport object associated with a specific media stream and its component (e.g., RTP or RTCP), facilitating direct manipulation or inspection of the transport.

```C
GstWebRTCICETransport *
gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
                               GstWebRTCICEStream * stream,
                               GstWebRTCICEComponent component)
```

```JavaScript
function GstWebRTC.WebRTCICE.prototype.find_transport(stream: GstWebRTC.WebRTCICEStream, component: GstWebRTC.WebRTCICEComponent): {
    // javascript wrapper for 'gst_webrtc_ice_find_transport'
}
```

```Python
def GstWebRTC.WebRTCICE.find_transport (self, stream, component):
    #python wrapper for 'gst_webrtc_ice_find_transport'
```

```APIDOC
GstWebRTC.WebRTCICE.find_transport(ice: GstWebRTCICE, stream: GstWebRTCICEStream, component: GstWebRTCICEComponent)
  Parameters:
    ice: The GstWebRTCICE instance
    stream: The GstWebRTCICEStream
    component: The GstWebRTCICEComponent
  Returns: GstWebRTCICETransport (or NULL/null/None on error)
  Since: 1.22
```

--------------------------------

### GstWebRTCICE Property: max-rtp-port

Source: https://gstreamer.freedesktop.org/documentation/webrtclib/gstwebrtcice.html?gi-language=c/gstwebrtcice

Defines the maximum port number for the local RTP port range. This value must be greater than or equal to 'min-rtp-port'.

```APIDOC
Property: max-rtp-port
  Type (C): guint
  Type (JS): Number
  Type (Python): int
  Description: Maximum port for local rtp port range. min-rtp-port must be <= max-rtp-port
  Flags: Read, Write, Construct
  Since: 1.20
```

```C
"max-rtp-port" guint
```

```JavaScript
"max-rtp-port" Number
```

```Python
"self.props.max_rtp_port" int
```
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