Major RTP fixes for using inbound SDP on outbound connection, get rid of

old local rtp stuff...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_dial.c

@@ -53,6 +53,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/utils.h"
 #include "asterisk/app.h"
 #include "asterisk/causes.h"
+#include "asterisk/rtp.h"
 #include "asterisk/manager.h"
 #include "asterisk/privacy.h"
 
@@ -310,7 +311,7 @@ static void hanguptree(struct localuser *outgoing, struct ast_channel *exception
 } while (0)
 
 
-static int onedigit_goto(struct ast_channel *chan, char *context, char exten, int pri) 
+static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri) 
 {
 	char rexten[2] = { exten, '\0' };
 
@@ -380,7 +381,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
 	int pos;
 	int single;
 	struct ast_channel *winner;
-	char *context = NULL;
+	const char *context = NULL;
 	char cidname[AST_MAX_EXTENSION];
 
 	single = (outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK));
@@ -475,6 +476,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
 						ast_clear_flag(o, DIAL_STILLGOING);	
 						HANDLE_CAUSE(cause, in);
 					} else {
+						ast_rtp_make_compatible(o->chan, in);
 						if (o->chan->cid.cid_num)
 							free(o->chan->cid.cid_num);
 						o->chan->cid.cid_num = NULL;
@@ -744,16 +746,17 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 	long timelimit = 0;
 	long play_warning = 0;
 	long warning_freq=0;
-	char *warning_sound=NULL;
-	char *end_sound=NULL;
-	char *start_sound=NULL;
+	const char *warning_sound=NULL;
+	const char *end_sound=NULL;
+	const char *start_sound=NULL;
 	char *dtmfcalled=NULL, *dtmfcalling=NULL;
-	char *var;
+	const char *var;
 	char status[256];
 	int play_to_caller=0,play_to_callee=0;
 	int sentringing=0, moh=0;
-	char *outbound_group = NULL;
-	char *macro_result = NULL, *macro_transfer_dest = NULL;
+	const char *outbound_group = NULL;
+	const char *macro_result = NULL;
+	char *macro_transfer_dest = NULL;
 	int digit = 0, result = 0;
 	time_t start_time, answer_time, end_time;
 	struct ast_app *app = NULL;
@@ -1052,6 +1055,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 			}
 		}
 
+		/* Setup outgoing SDP to match incoming one */
+		ast_rtp_make_compatible(tmp->chan, chan);
+		
 		/* Inherit specially named variables from parent channel */
 		ast_channel_inherit_variables(chan, tmp->chan);
 
@@ -1190,7 +1196,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 		if (peer->name)
 			pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
 
-		number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
+		number = (char *)pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
 		if (!number)
 			number = numsubst;
 		pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
@@ -1602,7 +1608,8 @@ static int dial_exec(struct ast_channel *chan, void *data)
 
 static int retrydial_exec(struct ast_channel *chan, void *data)
 {
-	char *announce = NULL, *context = NULL, *dialdata = NULL;
+	char *announce = NULL, *dialdata = NULL;
+	const char *context = NULL;
 	int sleep = 0, loops = 0, res = 0;
 	struct localuser *u;
 	struct ast_flags peerflags;

channels/chan_mgcp.c

@@ -2095,7 +2095,6 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	add_header(&resp, "X", sub->txident);
 	add_header(&resp, "I", sub->cxident);
 	/*add_header(&resp, "S", "");*/
-	ast_rtp_offered_from_local(sub->rtp, 0);
 	add_sdp(&resp, sub, rtp);
 	/* SC: fill in new fields */
 	resp.cmd = MGCP_CMD_MDCX;
@@ -2129,7 +2128,6 @@ static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	/* SC: X header should not be sent. kept for compatibility */
 	add_header(&resp, "X", sub->txident);
 	/*add_header(&resp, "S", "");*/
-	ast_rtp_offered_from_local(sub->rtp, 1);
 	add_sdp(&resp, sub, rtp);
 	/* SC: fill in new fields */
 	resp.cmd = MGCP_CMD_CRCX;
@@ -3948,7 +3946,7 @@ static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, stru
 	/* XXX Is there such thing as video support with MGCP? XXX */
 	struct mgcp_subchannel *sub;
 	sub = chan->tech_pvt;
-	if (sub) {
+	if (sub && !sub->alreadygone) {
 		transmit_modify_with_sdp(sub, rtp, codecs);
 		return 0;
 	}

channels/chan_sip.c

@@ -4509,7 +4509,6 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
 	}
 	respprep(&resp, p, msg, req);
 	if (p->rtp) {
-		ast_rtp_offered_from_local(p->rtp, 0);
 		add_sdp(&resp, p);
 	} else {
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
@@ -4583,7 +4582,6 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
 	add_header(&req, "Allow", ALLOWED_METHODS);
 	if (sipdebug)
 		add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
-	ast_rtp_offered_from_local(p->rtp, 1);
 	add_sdp(&req, p);
 	/* Use this as the basis */
 	copy_request(&p->initreq, &req);
@@ -4924,7 +4922,6 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
 		}
 	}
 	if (sdp && p->rtp) {
-		ast_rtp_offered_from_local(p->rtp, 1);
 		add_sdp(&req, p);
 	} else {
 		add_header_contentLength(&req, 0);
@@ -12691,6 +12688,11 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 	if (!p) 
 		return -1;
 	ast_mutex_lock(&p->lock);
+	if (ast_test_flag(p, SIP_ALREADYGONE)) {
+		/* If we're destroyed, don't bother */
+		ast_mutex_unlock(&p->lock);
+		return 0;
+	}
 	if (rtp)
 		ast_rtp_get_peer(rtp, &p->redirip);
 	else

frame.c

@@ -1299,3 +1299,28 @@ int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2)
 
 	return 0;
 }
+
+struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe)
+{
+	struct ast_frame *cur, *oldhead;
+	int len=0;
+	if (f && dupe)
+		f = ast_frdup(f);
+	if (!f)
+		return head;
+
+	f->next = NULL;
+	if (!head) 
+		return f;
+	cur = head;
+	while(cur->next) {
+		cur = cur->next;
+		len++;
+		if (len >= maxlen) {
+			oldhead = head;
+			head = head->next;
+			ast_frfree(oldhead);
+		}
+	}
+	return head;
+}

include/asterisk/frame.h

@@ -452,6 +452,10 @@ extern int ast_codec_get_samples(struct ast_frame *f);
 /*! \brief Returns the number of bytes for the number of samples of the given format */
 extern int ast_codec_get_len(int format, int samples);
 
+/*! \brief Appends a frame to the end of a list of frames, truncating the maximum length of the list */
+extern struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe);
+
+
 /*! \brief Gets duration in ms of interpolation frame for a format */
 static inline int ast_codec_interp_len(int format) 
 { 

include/asterisk/rtp.h

@@ -135,7 +135,6 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
 /*  Mapping between RTP payload format codes and Asterisk codes: */
 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
 int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
-void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local);
 
 void ast_rtp_get_current_formats(struct ast_rtp* rtp,
 			     int* astFormats, int* nonAstFormats);
@@ -154,6 +153,8 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
 
 void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
 
+int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src);
+
 void ast_rtp_stop(struct ast_rtp *rtp);
 
 void ast_rtp_init(void);

rtp.c

@@ -124,7 +124,6 @@ struct ast_rtp {
 	int rtp_lookup_code_cache_isAstFormat;
 	int rtp_lookup_code_cache_code;
 	int rtp_lookup_code_cache_result;
-	int rtp_offered_from_local;
 	struct ast_rtcp *rtcp;
 };
 
@@ -724,10 +723,98 @@ void ast_rtp_pt_default(struct ast_rtp* rtp)
 	rtp->rtp_lookup_code_cache_result = 0;
 }
 
+static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
+{
+	int i;
+	/* Copy payload types from source to destination */
+	for (i=0; i < MAX_RTP_PT; ++i) {
+		dest->current_RTP_PT[i].isAstFormat = 
+			src->current_RTP_PT[i].isAstFormat;
+		dest->current_RTP_PT[i].code = 
+			src->current_RTP_PT[i].code; 
+	}
+	dest->rtp_lookup_code_cache_isAstFormat = 0;
+	dest->rtp_lookup_code_cache_code = 0;
+	dest->rtp_lookup_code_cache_result = 0;
+}
+
+/*--- get_proto: Get channel driver interface structure */
+static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
+{
+	struct ast_rtp_protocol *cur;
+
+	cur = protos;
+	while(cur) {
+		if (cur->type == chan->type) {
+			return cur;
+		}
+		cur = cur->next;
+	}
+	return NULL;
+}
+
+int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
+{
+	struct ast_rtp *destp, *srcp;		/* Audio RTP Channels */
+	struct ast_rtp *vdestp, *vsrcp;		/* Video RTP channels */
+	struct ast_rtp_protocol *destpr, *srcpr;
+	/* Lock channels */
+	ast_mutex_lock(&dest->lock);
+	while(ast_mutex_trylock(&src->lock)) {
+		ast_mutex_unlock(&dest->lock);
+		usleep(1);
+		ast_mutex_lock(&dest->lock);
+	}
+
+	/* Find channel driver interfaces */
+	destpr = get_proto(dest);
+	srcpr = get_proto(src);
+	if (!destpr) {
+		ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
+		ast_mutex_unlock(&dest->lock);
+		ast_mutex_unlock(&src->lock);
+		return 0;
+	}
+	if (!srcpr) {
+		ast_log(LOG_WARNING, "Channel '%s' has no RTP, not doing anything\n", src->name);
+		ast_mutex_unlock(&dest->lock);
+		ast_mutex_unlock(&src->lock);
+		return 0;
+	}
+
+	/* Get audio and video interface (if native bridge is possible) */
+	destp = destpr->get_rtp_info(dest);
+	if (destpr->get_vrtp_info)
+		vdestp = destpr->get_vrtp_info(dest);
+	else
+		vdestp = NULL;
+	srcp = srcpr->get_rtp_info(src);
+	if (srcpr->get_vrtp_info)
+		vsrcp = srcpr->get_vrtp_info(src);
+	else
+		vsrcp = NULL;
+
+	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
+	if (!destp || !srcp) {
+		/* Somebody doesn't want to play... */
+		ast_mutex_unlock(&dest->lock);
+		ast_mutex_unlock(&src->lock);
+		return 0;
+	}
+	ast_rtp_pt_copy(destp, srcp);
+	if (vdestp && vsrcp)
+		ast_rtp_pt_copy(vdestp, vsrcp);
+	ast_mutex_unlock(&dest->lock);
+	ast_mutex_unlock(&src->lock);
+	ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
+	return 1;
+}
+
 /* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */
 /* By default, use the well-known value for this type (although it may */
 /* still be set to a different value by a subsequent "a=rtpmap:" line): */
-void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
+void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) 
+{
 	if (pt < 0 || pt > MAX_RTP_PT) 
 		return; /* bogus payload type */
 
@@ -739,7 +826,9 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
 /* Make a note of a RTP payload type (with MIME type) that was seen in */
 /* a SDP "a=rtpmap:" line. */
 void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
-			 char* mimeType, char* mimeSubtype) {
+			 char* mimeType, char* mimeSubtype) 
+			 
+{
 	int i;
 
 	if (pt < 0 || pt > MAX_RTP_PT) 
@@ -770,13 +859,6 @@ void ast_rtp_get_current_formats(struct ast_rtp* rtp,
 	}
 }
 
-void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
-	if (rtp)
-		rtp->rtp_offered_from_local = local;
-	else
-		ast_log(LOG_WARNING, "rtp structure is null\n");
-}
-
 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) 
 {
 	struct rtpPayloadType result;
@@ -786,8 +868,7 @@ struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
 		return result; /* bogus payload type */
 
 	/* Start with the negotiated codecs */
-	if (!rtp->rtp_offered_from_local)
-		result = rtp->current_RTP_PT[pt];
+	result = rtp->current_RTP_PT[pt];
 
 	/* If it doesn't exist, check our static RTP type list, just in case */
 	if (!result.code) 
@@ -829,7 +910,8 @@ int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int co
 	return -1;
 }
 
-char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) {
+char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) 
+{
 
 	int i;
 
@@ -1485,21 +1567,6 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
 	return 0;
 }
 
-/*--- get_proto: Get channel driver interface structure */
-static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
-{
-	struct ast_rtp_protocol *cur;
-
-	cur = protos;
-	while(cur) {
-		if (cur->type == chan->type) {
-			return cur;
-		}
-		cur = cur->next;
-	}
-	return NULL;
-}
-
 /* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate
 	re-invite so the peers exchange media directly outside 
 	of Asterisk. */
@@ -1698,11 +1765,11 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
 			*rc = who;
 			if (option_debug)
 				ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
-			if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) {
+			if ((c0->tech_pvt == pvt0)) {
 				if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) 
 					ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
 			}
-			if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) {
+			if ((c1->tech_pvt == pvt1)) {
 				if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) 
 					ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
 			}