webrtc.c File Reference

Client-side WebRTC signaling callback implementations. More...

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Functions
webrtc_signaling_callbacks_t	webrtc_get_signaling_callbacks (void)
	Get signaling callbacks for WebRTC peer manager.
void	webrtc_set_acds_transport (acip_transport_t *transport)
	Set the ACDS transport for signaling callbacks.
void	webrtc_set_session_context (const uint8_t session_id[16], const uint8_t participant_id[16])
	Set session and participant IDs for signaling.
void	webrtc_set_transport (acip_transport_t *transport)
	Set the WebRTC data channel transport.
acip_transport_t *	webrtc_get_transport (void)
	Get the current WebRTC data channel transport.
void	webrtc_cleanup_transport (void)
	Cleanup and release the WebRTC data channel transport.

Detailed Description

Client-side WebRTC signaling callback implementations.

Implements the send_sdp and send_ice callbacks that the peer manager uses to transmit local SDP/ICE to remote peers via ACDS.

PACKET CONSTRUCTION:

Both callbacks construct ACIP packets following the wire format:

• Fixed header (acip_webrtc_sdp_t or acip_webrtc_ice_t)
• Variable payload (SDP string or ICE candidate+mid)

THREAD SAFETY:

Callbacks use mutex protection for accessing shared state (ACDS transport, session context). Safe to call from peer manager threads.

Author

Zachary Fogg me@zf.nosp@m.o.gg

Date

January 2026

Definition in file src/client/webrtc.c.

Function Documentation

webrtc_cleanup_transport()

void webrtc_cleanup_transport

(

void

)

Cleanup and release the WebRTC data channel transport.

Closes and releases the WebRTC transport. Called when disconnecting or falling back to a different transport type.

Note

This is a convenience function that sets transport to NULL

Actual transport cleanup (closing sockets, freeing peer manager) should be done by the caller before calling this

Definition at line 365 of file src/client/webrtc.c.

                                    {
  ensure_mutex_initialized();
  mutex_lock(&g_signaling_mutex);
 
  if (g_webrtc_transport) {
    log_debug("Cleaning up WebRTC data channel transport");
    g_webrtc_transport = NULL;
  }
 
  mutex_unlock(&g_signaling_mutex);
}

webrtc_get_signaling_callbacks()

webrtc_signaling_callbacks_t webrtc_get_signaling_callbacks

(

void

)

Get signaling callbacks for WebRTC peer manager.

Returns a structure containing callback functions that send SDP/ICE via the ACDS TCP connection. The callbacks are stateless - they use the global ACDS transport to send signaling messages.

Returns

Signaling callbacks structure (valid until program exit)

Note

The returned structure points to static callbacks, safe to use for the lifetime of the peer manager.

Callbacks will fail if ACDS connection is not active.

Definition at line 265 of file src/client/webrtc.c.

                                                                  {
  ensure_mutex_initialized();
 
  webrtc_signaling_callbacks_t callbacks = {
      .send_sdp = client_send_sdp, .send_ice = client_send_ice, .user_data = NULL};
 
  return callbacks;
}

webrtc_get_transport()

acip_transport_t * webrtc_get_transport

(

void

)

Get the current WebRTC data channel transport.

Retrieves the active WebRTC transport for peer-to-peer communication.

Returns

Current WebRTC transport pointer (NULL if not set)

Note

Caller must not free the returned pointer

Pointer is valid only until webrtc_set_transport(NULL) is called

Definition at line 344 of file src/client/webrtc.c.

                                             {
  ensure_mutex_initialized();
  mutex_lock(&g_signaling_mutex);
 
  acip_transport_t *transport = g_webrtc_transport;
 
  mutex_unlock(&g_signaling_mutex);
 
  return transport;
}

webrtc_set_acds_transport()

void webrtc_set_acds_transport

(

acip_transport_t *

transport

)

Set the ACDS transport for signaling callbacks.

Configures the transport that will be used to send SDP/ICE messages. Must be called before peer manager generates any local descriptions.

Parameters

transport

ACDS TCP transport (NULL to clear)

Note

This is called automatically when connecting to ACDS.

Callbacks will fail with ERROR_INVALID_STATE if transport is NULL.

Definition at line 274 of file src/client/webrtc.c.

                                                            {
  ensure_mutex_initialized();
  mutex_lock(&g_signaling_mutex);
 
  g_acds_transport = transport;
 
  if (transport) {
    log_debug("ACDS transport set for WebRTC signaling");
  } else {
    log_debug("ACDS transport cleared for WebRTC signaling");
  }
 
  mutex_unlock(&g_signaling_mutex);
}

webrtc_set_session_context()

void webrtc_set_session_context	(	const uint8_t	session_id[16],
		const uint8_t	participant_id[16]
	)

Set session and participant IDs for signaling.

Configures the session context used when sending SDP/ICE messages. Must be called after successful ACDS session join.

Parameters

session_id	Session UUID (16 bytes, copied)
participant_id	Local participant UUID (16 bytes, copied)

Note

This is called automatically when joining ACDS session.

Callbacks will fail with ERROR_INVALID_STATE if IDs are not set.

Definition at line 289 of file src/client/webrtc.c.

                                                                                                {
  if (!session_id || !participant_id) {
    log_error("Invalid session context parameters");
    return;
  }
 
  ensure_mutex_initialized();
  mutex_lock(&g_signaling_mutex);
 
  memcpy(g_session_context.session_id, session_id, 16);
  memcpy(g_session_context.participant_id, participant_id, 16);
  g_session_context.is_set = true;
 
  log_info("Session context set for WebRTC signaling (session=%.8s..., participant=%.8s...)", (const char *)session_id,
           (const char *)participant_id);
 
  mutex_unlock(&g_signaling_mutex);
}

References participant_id, and session_id.

webrtc_set_transport()

void webrtc_set_transport

(

acip_transport_t *

transport

)

Set the WebRTC data channel transport.

Stores the active WebRTC transport that will be used for peer-to-peer communication. This is different from the ACDS signaling transport - it's the actual data channel.

Parameters

transport

WebRTC transport (NULL to clear)

Note

Called when WebRTC connection is established or when falling back to TCP

If NULL is passed, the transport is cleared (e.g., on disconnect)

Definition at line 319 of file src/client/webrtc.c.

                                                       {
  ensure_mutex_initialized();
  mutex_lock(&g_signaling_mutex);
 
  g_webrtc_transport = transport;
 
  if (transport) {
    log_debug("WebRTC data channel transport set");
  } else {
    log_debug("WebRTC data channel transport cleared");
  }
 
  mutex_unlock(&g_signaling_mutex);
}

Variable Documentation

is_set

bool is_set

Definition at line 64 of file src/client/webrtc.c.

Referenced by options_config_validate().

participant_id

uint8_t participant_id[16]

Definition at line 63 of file src/client/webrtc.c.

Referenced by consensus_election_verify(), database_session_join(), database_session_leave(), nat_quality_to_acip(), negotiate_init(), session_host_add_memory_participant(), session_host_inject_audio(), session_host_inject_frame(), webrtc_peer_manager_connect(), webrtc_peer_manager_handle_sdp(), and webrtc_set_session_context().

session_id

uint8_t session_id[16]

Definition at line 62 of file src/client/webrtc.c.

Referenced by consensus_create_for_discovery(), database_session_clear_host(), database_session_create(), database_session_find_by_id(), database_session_is_migration_ready(), database_session_leave(), database_session_start_migration(), database_session_update_host(), nat_quality_to_acip(), negotiate_init(), signaling_broadcast(), turn_generate_credentials(), webrtc_peer_manager_connect(), and webrtc_set_session_context().