This patch for r425921 introduced a different 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, particularly in 12:
(1) The format capabilities structures and how they are used are a bit harder
to manipulate than they are in 13
(2) 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/branches/12@425943
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
struct ast_format format;
RAII_VAR(struct ast_format_cap *, caps, NULL, ast_format_cap_destroy);
enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
+
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
!ast_format_cap_is_empty(session->direct_media_cap);
+
int use_override_prefs = session->override_prefs.formats[0].id;
struct ast_codec_pref *prefs = use_override_prefs ?
&session->override_prefs : &session->endpoint->media.prefs;
if ((use_override_prefs && !codec_pref_has_type(&session->override_prefs, media_type)) ||
- (!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;