Don't call close_pc in SIP plugin if there was no SDP (fixes #3332) (#…

…3339)
meetecho · Mar 25, 2024 · 0f32c32 · 0f32c32
1 parent 5c6b2bb
commit 0f32c32
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Showing 3 changed files with 16 additions and 7 deletions.
diff --git a/src/ice.c b/src/ice.c
@@ -1411,6 +1411,7 @@ janus_ice_handle *janus_ice_handle_create(void *core_session, const char *opaque
 	handle->queued_candidates = g_async_queue_new();
 	handle->queued_packets = g_async_queue_new();
 	janus_mutex_init(&handle->mutex);
+	janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT);
 	janus_session_handles_insert(session, handle);
 	return handle;
 }

diff --git a/src/plugins/janus_sip.c b/src/plugins/janus_sip.c
@@ -696,8 +696,8 @@
 
 
 /* Plugin information */
-#define JANUS_SIP_VERSION			8
-#define JANUS_SIP_VERSION_STRING	"0.0.8"
+#define JANUS_SIP_VERSION			9
+#define JANUS_SIP_VERSION_STRING	"0.0.9"
 #define JANUS_SIP_DESCRIPTION		"This is a simple SIP plugin for Janus, allowing WebRTC peers to register at a SIP server and call SIP user agents through a Janus instance."
 #define JANUS_SIP_NAME				"JANUS SIP plugin"
 #define JANUS_SIP_AUTHOR			"Meetecho s.r.l."
@@ -2466,7 +2466,7 @@ void janus_sip_setup_media(janus_plugin_session *handle) {
 	g_atomic_int_set(&session->establishing, 0);
 	g_atomic_int_set(&session->hangingup, 0);
 	janus_mutex_unlock(&sessions_mutex);
-	/* TODO Only relay RTP/RTCP when we get this event */
+	/* Only relay RTP/RTCP when we get this event */
 }
 
 void janus_sip_incoming_rtp(janus_plugin_session *handle, janus_plugin_rtp *packet) {
@@ -4286,7 +4286,7 @@ static void *janus_sip_handler(void *data) {
 			if(session->stack->s_nh_i == NULL) {
 				JANUS_LOG(LOG_WARN, "NUA Handle for 200 OK still null??\n");
 			}
-			int response_code = 486;
+			int response_code = 603;
 			json_t *code_json = json_object_get(root, "code");
 			if(code_json)
 				response_code = json_integer_value(code_json);
@@ -5141,8 +5141,15 @@ void janus_sip_sofia_callback(nua_event_t event, int status, char const *phrase,
 				session->hangup_reason_header = NULL;
 				session->hangup_reason_header_protocol = NULL;
 				session->hangup_reason_header_cause = NULL;
-				if(g_atomic_int_get(&session->establishing) || g_atomic_int_get(&session->established))
-					gateway->close_pc(session->handle);
+				if(g_atomic_int_get(&session->establishing) || g_atomic_int_get(&session->established)) {
+					if(session->media.has_audio || session->media.has_video) {
+						/* Get rid of the PeerConnection in the core */
+						gateway->close_pc(session->handle);
+					} else {
+						/* No SDP was exchanged, just clean up locally */
+						janus_sip_hangup_media_internal(session->handle);
+					}
+				}
 			} else if(session->stack->s_nh_i == nh && callstate == nua_callstate_calling && session->status == janus_sip_call_status_incall) {
 				/* Have just sent re-INVITE */
 				janus_sip_call_update_status(session, janus_sip_call_status_incall_reinviting);

diff --git a/src/plugins/plugin.h b/src/plugins/plugin.h
@@ -401,7 +401,8 @@ struct janus_callbacks {
 
 	/*! \brief Callback to ask the core to close a WebRTC PeerConnection
 	 * \note A call to this method will result in the core invoking the hangup_media
-	 * callback on this plugin when done
+	 * callback on this plugin when done, but only if a PeerConnection had been
+	 * created or was in the process of being negotiated (SDP exchanged)
 	 * @param[in] handle The plugin/gateway session that the PeerConnection is related to */
 	void (* const close_pc)(janus_plugin_session *handle);
 	/*! \brief Callback to ask the core to get rid of a plugin/gateway session