diff options
author | Matthew Jordan <mjordan@digium.com> | 2014-10-19 04:03:35 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2014-10-19 04:03:35 +0000 |
commit | 404b6ab3ab5d0ea256401af257d91f899cc4b9fa (patch) | |
tree | f4318b8276431cf2b5fcbecc22df025bde295f1a /res/res_pjsip_sdp_rtp.c | |
parent | b263c8bdae3c3e40cc8d7d103af47f28ee1a9cec (diff) |
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
Diffstat (limited to 'res/res_pjsip_sdp_rtp.c')
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 6 |
1 files changed, 4 insertions, 2 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 092157cbc..1f863008f 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/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)) { |