[asterisk-commits] mmichelson: branch mmichelson/direct_media r383033 - in /team/mmichelson/dire...

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Wed Mar 13 11:14:47 2013
New Revision: 383033

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=383033
Log:
Put in changes so that proper codecs are placed in direct media SDPs.


Modified:
    team/mmichelson/direct_media/include/asterisk/res_sip_session.h
    team/mmichelson/direct_media/res/res_sip_sdp_audio.c
    team/mmichelson/direct_media/res/res_sip_session.c

Modified: team/mmichelson/direct_media/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/direct_media/include/asterisk/res_sip_session.h?view=diff&rev=383033&r1=383032&r2=383033
==============================================================================
--- team/mmichelson/direct_media/include/asterisk/res_sip_session.h (original)
+++ team/mmichelson/direct_media/include/asterisk/res_sip_session.h Wed Mar 13 11:14:47 2013
@@ -100,6 +100,8 @@
 	AST_LIST_HEAD_NOLOCK(, ast_sip_session_delayed_request) delayed_requests;
 	/* When we need to reschedule a reinvite, we use this structure to do it */
 	pj_timer_entry rescheduled_reinvite;
+	/* Format capabilities pertaining to direct media */
+	struct ast_format_cap *direct_media_cap;
 };
 
 typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);

Modified: team/mmichelson/direct_media/res/res_sip_sdp_audio.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/direct_media/res/res_sip_sdp_audio.c?view=diff&rev=383033&r1=383032&r2=383033
==============================================================================
--- team/mmichelson/direct_media/res/res_sip_sdp_audio.c (original)
+++ team/mmichelson/direct_media/res/res_sip_sdp_audio.c Wed Mar 13 11:14:47 2013
@@ -148,6 +148,15 @@
 	pj_str_t stmp;
 	pjmedia_sdp_attr *attr;
 	int index = 0, min_packet_size = 0, noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0;
+	RAII_VAR(struct ast_format_cap *, cap, ast_format_cap_alloc_nolock(), ast_format_cap_destroy);
+	int direct_media_enabled = !ast_sockaddr_isnull(&session->media[AST_SIP_MEDIA_AUDIO].direct_media_addr) &&
+		!ast_format_cap_is_empty(session->direct_media_cap);
+
+	if (direct_media_enabled) {
+		ast_format_cap_joint_copy(session->endpoint->codecs, session->direct_media_cap, cap);
+	} else {
+		ast_format_cap_copy(cap, session->endpoint->codecs);
+	}
 
 	if (!ast_format_cap_has_type(session->endpoint->codecs, AST_FORMAT_TYPE_AUDIO)) {
 		/* If no audio formats are configured don't add a stream */
@@ -166,7 +175,7 @@
 	media->desc.transport = STR_RTP_AVP;
 
 	/* Add connection level details */
-	if (!ast_sockaddr_isnull(&session->media[AST_SIP_MEDIA_AUDIO].direct_media_addr)) {
+	if (direct_media_enabled) {
 		ast_sockaddr_copy(&addr, &session->media[AST_SIP_MEDIA_AUDIO].direct_media_addr);
 	} else {
 		ast_rtp_instance_get_local_address(session->media[AST_SIP_MEDIA_AUDIO].rtp, &addr);
@@ -180,6 +189,7 @@
 	/* Add formats */
 	for (index = 0; (index < AST_CODEC_PREF_SIZE); index++) {
 		struct ast_format format;
+		struct ast_format compat_format;
 		int rtp_code;
 		pjmedia_sdp_rtpmap rtpmap;
 		struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(session->media[AST_SIP_MEDIA_AUDIO].rtp)->pref;
@@ -188,8 +198,10 @@
 			break;
 		} else if (AST_FORMAT_GET_TYPE(format.id) != AST_FORMAT_TYPE_AUDIO) {
 			continue;
+		} else if (!ast_format_cap_get_compatible_format(cap, &format, &compat_format)) {
+			continue;
 		} else if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session->media[AST_SIP_MEDIA_AUDIO].rtp),
-								   1, &format, 0)) == -1) {
+								   1, &compat_format, 0)) == -1) {
 			return -1;
 		}
 
@@ -269,6 +281,8 @@
 	RAII_VAR(struct ast_format_cap *, jointcap, NULL, ast_format_cap_destroy);
 	RAII_VAR(struct ast_format_cap *, peercap, NULL, ast_format_cap_destroy);
 	struct ast_format fmt;
+	int direct_media_enabled = !ast_sockaddr_isnull(&session->media[AST_SIP_MEDIA_AUDIO].direct_media_addr) &&
+		!ast_format_cap_is_empty(session->direct_media_cap);
 
 	/* Create an RTP instance if need be */
 	if (!session->media[AST_SIP_MEDIA_AUDIO].rtp && audio_create_rtp(session, session->endpoint->rtp_ipv6)) {
@@ -341,7 +355,11 @@
 	}
 
 	/* Using the configured codecs and the codecs in this SDP we determine the joint formats for *audio only* */
-	ast_format_cap_copy(cap, session->endpoint->codecs);
+	if (direct_media_enabled) {
+		ast_format_cap_joint_copy(session->endpoint->codecs, session->direct_media_cap, cap);
+	} else {
+		ast_format_cap_copy(cap, session->endpoint->codecs);
+	}
 	ast_format_cap_remove_bytype(cap, AST_FORMAT_TYPE_VIDEO);
 
 	if (!(jointcap = ast_format_cap_joint(cap, peercap))) {

Modified: team/mmichelson/direct_media/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/direct_media/res/res_sip_session.c?view=diff&rev=383033&r1=383032&r2=383033
==============================================================================
--- team/mmichelson/direct_media/res/res_sip_session.c (original)
+++ team/mmichelson/direct_media/res/res_sip_session.c Wed Mar 13 11:14:47 2013
@@ -681,6 +681,7 @@
 			iter->session_begin(session);
 		}
 	}
+	session->direct_media_cap = ast_format_cap_alloc_nolock();
 	inv_session->mod_data[session_module.id] = session;
 	AST_LIST_HEAD_INIT_NOLOCK(&session->delayed_requests);
 	ao2_ref(session, +1);