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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, 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/branches/13@425944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -899,11 +899,13 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
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int rtp_code;
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RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
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enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
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int use_override_prefs = ast_format_cap_count(session->req_caps);
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int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
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ast_format_cap_count(session->direct_media_cap);
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if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type) ||
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!ast_format_cap_has_type(session->req_caps, media_type)) {
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if ((use_override_prefs && !ast_format_cap_has_type(session->req_caps, media_type)) ||
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(!use_override_prefs && !ast_format_cap_has_type(session->endpoint->media.codecs, media_type))) {
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/* If no type formats are configured don't add a stream */
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return 0;
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} else if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->media.rtp.ipv6)) {
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