[asterisk-commits] mjordan: branch 13 r425944 - /branches/13/res/res_pjsip_sdp_rtp.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sat Oct 18 23:01:34 CDT 2014
Author: mjordan
Date: Sat Oct 18 23:01:31 2014
New Revision: 425944
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=425944
Log:
res/res_pjsip_sdp_rtp: Revert 425922
This patch for r425922 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,
Itch 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, 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.
Modified:
branches/13/res/res_pjsip_sdp_rtp.c
Modified: branches/13/res/res_pjsip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/res/res_pjsip_sdp_rtp.c?view=diff&rev=425944&r1=425943&r2=425944
==============================================================================
--- branches/13/res/res_pjsip_sdp_rtp.c (original)
+++ branches/13/res/res_pjsip_sdp_rtp.c Sat Oct 18 23:01:31 2014
@@ -899,11 +899,13 @@
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)) {
More information about the asterisk-commits
mailing list