res/res_pjsip_sdp_rtp: Revert 425924

This patch for r425924 introduced a bug, wherein sending an INVITE request
with no SDP would cause Asterisk to not send an SDP Offer in the 200
OK. The current structure of res_pjsip_sdp_rtp is a bit hard to deal with
to fix this, as create_outgoing_sdp has no knowledge of whether or not it is
creating an SDP as a new Offer or an Answer. This is something of an oversight
in the callback definition, as the caller of it does have this information.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res/res_pjsip_sdp_rtp.c

@@ -899,11 +899,13 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 	int rtp_code;
 	RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
 	enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
+	int use_override_prefs = ast_format_cap_count(session->req_caps);
+
 	int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
 		ast_format_cap_count(session->direct_media_cap);
 
-	if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type) ||
-	    !ast_format_cap_has_type(session->req_caps, media_type)) {
+	if ((use_override_prefs && !ast_format_cap_has_type(session->req_caps, media_type)) ||
+	    (!use_override_prefs && !ast_format_cap_has_type(session->endpoint->media.codecs, media_type))) {
 		/* If no type formats are configured don't add a stream */
 		return 0;
 	} else if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->media.rtp.ipv6)) {