static gboolean
google_session_handle_initiate(JabberStream *js, GoogleSession *session, xmlnode *sess, const char *iq_id)
{
const gchar *xmlns;
GoogleAVSessionData *session_data =
(GoogleAVSessionData *) session->session_data;
if (session->state != UNINIT) {
purple_debug_error("jabber", "Received initiate for active session.\n");
return FALSE;
}
session->description = xmlnode_copy(xmlnode_get_child(sess, "description"));
xmlns = xmlnode_get_namespace(session->description);
if (purple_strequal(xmlns, NS_GOOGLE_SESSION_PHONE))
session_data->video = FALSE;
else if (purple_strequal(xmlns, NS_GOOGLE_SESSION_VIDEO))
session_data->video = TRUE;
else {
purple_debug_error("jabber", "Received initiate with "
"invalid namespace %s.\n", xmlns);
return FALSE;
}
session_data->media = purple_media_manager_create_media(
purple_media_manager_get(),
purple_connection_get_account(js->gc),
"fsrtpconference", session->remote_jid, FALSE);
purple_media_set_prpl_data(session_data->media, session);
g_signal_connect_swapped(G_OBJECT(session_data->media),
"candidates-prepared",
G_CALLBACK(google_session_ready), session);
g_signal_connect_swapped(G_OBJECT(session_data->media), "codecs-changed",
G_CALLBACK(google_session_ready), session);
g_signal_connect(G_OBJECT(session_data->media), "state-changed",
G_CALLBACK(google_session_state_changed_cb), session);
g_signal_connect(G_OBJECT(session_data->media), "stream-info",
G_CALLBACK(google_session_stream_info_cb), session);
session->iq_id = g_strdup(iq_id);
if (js->google_relay_host && js->google_relay_token) {
jabber_google_do_relay_request(js, session,
jabber_google_relay_response_session_handle_initiate_cb);
} else {
jabber_google_relay_response_session_handle_initiate_cb(session, NULL,
0, 0, 0, NULL, NULL);
}
return TRUE;
}
static void
jabber_google_relay_response_session_initiate_cb(GoogleSession *session,
const gchar *relay_ip, guint relay_udp, guint relay_tcp, guint relay_ssltcp,
const gchar *relay_username, const gchar *relay_password)
{
GParameter *params;
guint num_params;
JabberStream *js = session->js;
GoogleAVSessionData *session_data =
(GoogleAVSessionData *) session->session_data;
session_data->media = purple_media_manager_create_media(
purple_media_manager_get(),
purple_connection_get_account(js->gc),
"fsrtpconference", session->remote_jid, TRUE);
purple_media_set_prpl_data(session_data->media, session);
g_signal_connect_swapped(G_OBJECT(session_data->media),
"candidates-prepared",
G_CALLBACK(google_session_ready), session);
g_signal_connect_swapped(G_OBJECT(session_data->media), "codecs-changed",
G_CALLBACK(google_session_ready), session);
g_signal_connect(G_OBJECT(session_data->media), "state-changed",
G_CALLBACK(google_session_state_changed_cb), session);
g_signal_connect(G_OBJECT(session_data->media), "stream-info",
G_CALLBACK(google_session_stream_info_cb), session);
params =
jabber_google_session_get_params(js, relay_ip, relay_udp, relay_tcp,
relay_ssltcp, relay_username, relay_password, &num_params);
if (purple_media_add_stream(session_data->media, "google-voice",
session->remote_jid, PURPLE_MEDIA_AUDIO,
TRUE, "nice", num_params, params) == FALSE ||
(session_data->video && purple_media_add_stream(
session_data->media, "google-video",
session->remote_jid, PURPLE_MEDIA_VIDEO,
TRUE, "nice", num_params, params) == FALSE)) {
purple_media_error(session_data->media, "Error adding stream.");
purple_media_end(session_data->media, NULL, NULL);
g_free(params);
} else {
session_data->added_streams = TRUE;
}
g_free(params);
}
/*------------------------------------------------------------------------
* Initiate a voice/video session with a contact.
*
* @param account The MXit account object
* @param who The username of the contact.
* @param type The type of media session to initiate
* @return TRUE if session was initiated
*/
gboolean mxit_media_initiate(PurpleAccount *account, const char *who, PurpleMediaSessionType type)
{
gchar* transmitter = "rawudp";
PurpleMedia* media = NULL;
purple_debug_info(MXIT_PLUGIN_ID, "mxit_media_initiate: buddy '%s'\n", who);
media = purple_media_manager_create_media(
purple_media_manager_get(),
account,
"fsrtpconference",
who,
TRUE
);
if (!media) {
purple_debug_info(MXIT_PLUGIN_ID, "mxit_media_initiate: could not create media session\n");
return FALSE;
}
/* attach callbacks */
g_signal_connect(G_OBJECT(media), "candidates-prepared", G_CALLBACK(mxit_candidates_prepared_cb), NULL);
g_signal_connect(G_OBJECT(media), "stream-info", G_CALLBACK(mxit_stream_info_cb), NULL);
g_signal_connect(G_OBJECT(media), "state-changed", G_CALLBACK(mxit_state_changed_cb), NULL);
/* initiate audio session */
if ((type & PURPLE_MEDIA_AUDIO) &&
(!purple_media_add_stream(media, "audio", who, PURPLE_MEDIA_AUDIO, TRUE, transmitter, 0, NULL))) {
purple_media_end(media, NULL, NULL);
return FALSE;
}
/* initiate video session */
if ((type & PURPLE_MEDIA_VIDEO) &&
(!purple_media_add_stream(media, "video", who, PURPLE_MEDIA_VIDEO, TRUE, transmitter, 0, NULL))) {
purple_media_end(media, NULL, NULL);
return FALSE;
}
return TRUE;
}
struct sipe_backend_media *
sipe_backend_media_new(struct sipe_core_public *sipe_public,
struct sipe_media_call *call,
const gchar *participant,
gboolean initiator)
{
struct sipe_backend_media *media = g_new0(struct sipe_backend_media, 1);
struct sipe_backend_private *purple_private = sipe_public->backend_private;
PurpleMediaManager *manager = purple_media_manager_get();
GstElement *pipeline;
media->m = purple_media_manager_create_media(manager,
purple_private->account,
"fsrtpconference",
participant, initiator);
g_signal_connect(G_OBJECT(media->m), "candidates-prepared",
G_CALLBACK(on_candidates_prepared_cb), call);
g_signal_connect(G_OBJECT(media->m), "codecs-changed",
G_CALLBACK(on_codecs_changed_cb), call);
g_signal_connect(G_OBJECT(media->m), "stream-info",
G_CALLBACK(on_stream_info_cb), call);
g_signal_connect(G_OBJECT(media->m), "error",
G_CALLBACK(on_error_cb), call);
g_signal_connect(G_OBJECT(media->m), "state-changed",
G_CALLBACK(on_state_changed_cb), call);
/* On error, the pipeline is no longer in PLAYING state and libpurple
* will not switch it back to PLAYING, preventing any more calls until
* application restart. We switch the state ourselves here to negate
* effect of any error in previous call (if any). */
pipeline = purple_media_manager_get_pipeline(manager);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
return media;
}
