Merge in the RTP engine API.

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3

UPGRADE.txt

@@ -20,7 +20,11 @@
 
 From 1.6.2 to 1.6.3:
 
-* Nothing, yet!
+* The usage of RTP inside of Asterisk has now become modularized. This means
+  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
+  If you are not using autoload=yes in modules.conf you will need to ensure
+  it is set to load. If not, then any module which uses RTP (such as chan_sip)
+  will not be able to send or receive calls.
 
 From 1.6.1 to 1.6.2:
 

apps/app_dial.c

@@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/utils.h"
 #include "asterisk/app.h"
 #include "asterisk/causes.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/cdr.h"
 #include "asterisk/manager.h"
 #include "asterisk/privacy.h"
@@ -745,7 +745,9 @@ static void do_forward(struct chanlist *o,
 		char *new_cid_num, *new_cid_name;
 		struct ast_channel *src;
 
-		ast_rtp_make_compatible(c, in, single);
+		if (single) {
+			ast_rtp_instance_early_bridge_make_compatible(c, in);
+		}
 		if (ast_test_flag64(o, OPT_FORCECLID)) {
 			new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
 			new_cid_name = NULL; /* XXX no name ? */
@@ -1745,7 +1747,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
 		pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
 
 		/* Setup outgoing SDP to match incoming one */
-		ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
+		if (!outgoing && !rest) {
+			ast_rtp_instance_early_bridge_make_compatible(tc, chan);
+		}
 		
 		/* Inherit specially named variables from parent channel */
 		ast_channel_inherit_variables(chan, tc);

channels/chan_agent.c

@@ -52,7 +52,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/file.h"

channels/chan_bridge.c

@@ -39,7 +39,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/file.h"

channels/chan_gtalk.c

@@ -52,7 +52,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/stun.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/file.h"
@@ -112,8 +113,8 @@ struct gtalk_pvt {
 	char cid_name[80];               /*!< Caller ID name */
 	char exten[80];                  /*!< Called extension */
 	struct ast_channel *owner;       /*!< Master Channel */
-	struct ast_rtp *rtp;             /*!< RTP audio session */
-	struct ast_rtp *vrtp;            /*!< RTP video session */
+	struct ast_rtp_instance *rtp;             /*!< RTP audio session */
+	struct ast_rtp_instance *vrtp;            /*!< RTP video session */
 	int jointcapability;             /*!< Supported capability at both ends (codecs ) */
 	int peercapability;
 	struct gtalk_pvt *next;	/* Next entity */
@@ -183,11 +184,6 @@ static int gtalk_sendhtml(struct ast_channel *ast, int subclass, const char *dat
 static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const char *them, const char *sid);
 static char *gtalk_do_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
 static char *gtalk_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-/*----- RTP interface functions */
-static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
-							   struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
-static enum ast_rtp_get_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static int gtalk_get_codec(struct ast_channel *chan);
 
 /*! \brief PBX interface structure for channel registration */
 static const struct ast_channel_tech gtalk_tech = {
@@ -197,7 +193,7 @@ static const struct ast_channel_tech gtalk_tech = {
 	.requester = gtalk_request,
 	.send_digit_begin = gtalk_digit_begin,
 	.send_digit_end = gtalk_digit_end,
-	.bridge = ast_rtp_bridge,
+	.bridge = ast_rtp_instance_bridge,
 	.call = gtalk_call,
 	.hangup = gtalk_hangup,
 	.answer = gtalk_answer,
@@ -216,14 +212,6 @@ static struct sched_context *sched;	/*!< The scheduling context */
 static struct io_context *io;	/*!< The IO context */
 static struct in_addr __ourip;
 
-/*! \brief RTP driver interface */
-static struct ast_rtp_protocol gtalk_rtp = {
-	type: "Gtalk",
-	get_rtp_info: gtalk_get_rtp_peer,
-	set_rtp_peer: gtalk_set_rtp_peer,
-	get_codec: gtalk_get_codec,
-};
-
 static struct ast_cli_entry gtalk_cli[] = {
 	AST_CLI_DEFINE(gtalk_do_reload, "Reload GoogleTalk configuration"),
 	AST_CLI_DEFINE(gtalk_show_channels, "Show GoogleTalk channels"),
@@ -371,7 +359,7 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
 		iks_insert_node(dcodecs, payload_gsm);
 		res++;
 	}
-	ast_rtp_lookup_code(p->rtp, 1, codec);
+
 	return res;
 }
 
@@ -523,18 +511,19 @@ static int gtalk_answer(struct ast_channel *ast)
 	return res;
 }
 
-static enum ast_rtp_get_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result gtalk_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
 	struct gtalk_pvt *p = chan->tech_pvt;
-	enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
+	enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
 
 	if (!p)
 		return res;
 
 	ast_mutex_lock(&p->lock);
 	if (p->rtp){
-		*rtp = p->rtp;
-		res = AST_RTP_TRY_PARTIAL;
+		ao2_ref(p->rtp, +1);
+		*instance = p->rtp;
+		res = AST_RTP_GLUE_RESULT_LOCAL;
 	}
 	ast_mutex_unlock(&p->lock);
 
@@ -547,7 +536,7 @@ static int gtalk_get_codec(struct ast_channel *chan)
 	return p->peercapability;
 }
 
-static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
 {
 	struct gtalk_pvt *p;
 
@@ -567,6 +556,13 @@ static int gtalk_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, str
 	return 0;
 }
 
+static struct ast_rtp_glue gtalk_rtp_glue = {
+	.type = "Gtalk",
+	.get_rtp_info = gtalk_get_rtp_peer,
+	.get_codec = gtalk_get_codec,
+	.update_peer = gtalk_set_rtp_peer,
+};
+
 static int gtalk_response(struct gtalk *client, char *from, ikspak *pak, const char *reasonstr, const char *reasonstr2)
 {
 	iks *response = NULL, *error = NULL, *reason = NULL;
@@ -617,13 +613,13 @@ static int gtalk_is_answered(struct gtalk *client, ikspak *pak)
 	/* codec points to the first <payload-type/> tag */
 	codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
 	while (codec) {
-		ast_rtp_set_m_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")));
-		ast_rtp_set_rtpmap_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+		ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")));
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
 		codec = iks_next_tag(codec);
 	}
 	
 	/* Now gather all of the codecs that we are asked for */
-	ast_rtp_get_current_formats(tmp->rtp, &tmp->peercapability, &peernoncodeccapability);
+	ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(tmp->rtp), &tmp->peercapability, &peernoncodeccapability);
 	
 	/* at this point, we received an awser from the remote Gtalk client,
 	   which allows us to compare capabilities */
@@ -810,7 +806,7 @@ static int gtalk_create_candidates(struct gtalk *client, struct gtalk_pvt *p, ch
 		goto safeout;
 	}
 
-	ast_rtp_get_us(p->rtp, &sin);
+	ast_rtp_instance_get_local_address(p->rtp, &sin);
 	ast_find_ourip(&us, bindaddr);
 	if (!strcmp(ast_inet_ntoa(us), "127.0.0.1")) {
 		ast_log(LOG_WARNING, "Found a loopback IP on the system, check your network configuration or set the bindaddr attribute.");
@@ -951,8 +947,9 @@ static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const
 		tmp->initiator = 1;
 	}
 	/* clear codecs */
-	tmp->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
-	ast_rtp_pt_clear(tmp->rtp);
+	tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+	ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_RTCP, 1);
+	ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp);
 
 	/* add user configured codec capabilites */
 	if (client->capability)
@@ -1014,20 +1011,20 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
 
 	/* Set Frame packetization */
 	if (i->rtp)
-		ast_rtp_codec_setpref(i->rtp, &i->prefs);
+		ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
 
 	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
 	fmt = ast_best_codec(tmp->nativeformats);
 
 	if (i->rtp) {
-		ast_rtp_setstun(i->rtp, 1);
-		ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
-		ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
+		ast_rtp_instance_set_prop(i->rtp, AST_RTP_PROPERTY_STUN, 1);
+		ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
+		ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
 	}
 	if (i->vrtp) {
-		ast_rtp_setstun(i->rtp, 1);
-		ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
-		ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
+		ast_rtp_instance_set_prop(i->vrtp, AST_RTP_PROPERTY_STUN, 1);
+		ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
+		ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
 	}
 	if (state == AST_STATE_RING)
 		tmp->rings = 1;
@@ -1142,9 +1139,9 @@ static void gtalk_free_pvt(struct gtalk *client, struct gtalk_pvt *p)
 	if (p->owner)
 		ast_log(LOG_WARNING, "Uh oh, there's an owner, this is going to be messy.\n");
 	if (p->rtp)
-		ast_rtp_destroy(p->rtp);
+		ast_rtp_instance_destroy(p->rtp);
 	if (p->vrtp)
-		ast_rtp_destroy(p->vrtp);
+		ast_rtp_instance_destroy(p->vrtp);
 	gtalk_free_candidates(p->theircandidates);
 	ast_free(p);
 }
@@ -1207,13 +1204,13 @@ static int gtalk_newcall(struct gtalk *client, ikspak *pak)
 	codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
 	
 	while (codec) {
-		ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
-		ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+		ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
 		codec = iks_next_tag(codec);
 	}
 	
 	/* Now gather all of the codecs that we are asked for */
-	ast_rtp_get_current_formats(p->rtp, &p->peercapability, &peernoncodeccapability);
+	ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(p->rtp), &p->peercapability, &peernoncodeccapability);
 	p->jointcapability = p->capability & p->peercapability;
 	ast_mutex_unlock(&p->lock);
 		
@@ -1277,16 +1274,16 @@ static int gtalk_update_stun(struct gtalk *client, struct gtalk_pvt *p)
 			 p->ourcandidates->username);
 		
 		/* Find out the result of the STUN */
-		ast_rtp_get_peer(p->rtp, &aux);
+		ast_rtp_instance_get_remote_address(p->rtp, &aux);
 
 		/* If the STUN result is different from the IP of the hostname,
 			lock on the stun IP of the hostname advertised by the
 			remote client */
 		if (aux.sin_addr.s_addr && 
 		    aux.sin_addr.s_addr != sin.sin_addr.s_addr)
-			ast_rtp_stun_request(p->rtp, &aux, username);
+			ast_rtp_instance_stun_request(p->rtp, &aux, username);
 		else 
-			ast_rtp_stun_request(p->rtp, &sin, username);
+			ast_rtp_instance_stun_request(p->rtp, &sin, username);
 		
 		if (aux.sin_addr.s_addr) {
 			ast_debug(4, "Receiving RTP traffic from IP %s, matches with remote candidate's IP %s\n", ast_inet_ntoa(aux.sin_addr), tmp->ip);
@@ -1387,7 +1384,7 @@ static struct ast_frame *gtalk_rtp_read(struct ast_channel *ast, struct gtalk_pv
 
 	if (!p->rtp)
 		return &ast_null_frame;
-	f = ast_rtp_read(p->rtp);
+	f = ast_rtp_instance_read(p->rtp, 0);
 	gtalk_update_stun(p->parent, p);
 	if (p->owner) {
 		/* We already hold the channel lock */
@@ -1438,7 +1435,7 @@ static int gtalk_write(struct ast_channel *ast, struct ast_frame *frame)
 		if (p) {
 			ast_mutex_lock(&p->lock);
 			if (p->rtp) {
-				res = ast_rtp_write(p->rtp, frame);
+				res = ast_rtp_instance_write(p->rtp, frame);
 			}
 			ast_mutex_unlock(&p->lock);
 		}
@@ -1447,7 +1444,7 @@ static int gtalk_write(struct ast_channel *ast, struct ast_frame *frame)
 		if (p) {
 			ast_mutex_lock(&p->lock);
 			if (p->vrtp) {
-				res = ast_rtp_write(p->vrtp, frame);
+				res = ast_rtp_instance_write(p->vrtp, frame);
 			}
 			ast_mutex_unlock(&p->lock);
 		}
@@ -2062,7 +2059,7 @@ static int load_module(void)
 		return 0;
 	}
 
-	ast_rtp_proto_register(&gtalk_rtp);
+	ast_rtp_glue_register(&gtalk_rtp_glue);
 	ast_cli_register_multiple(gtalk_cli, ARRAY_LEN(gtalk_cli));
 
 	/* Make sure we can register our channel type */
@@ -2086,7 +2083,7 @@ static int unload_module(void)
 	ast_cli_unregister_multiple(gtalk_cli, ARRAY_LEN(gtalk_cli));
 	/* First, take us out of the channel loop */
 	ast_channel_unregister(&gtalk_tech);
-	ast_rtp_proto_unregister(&gtalk_rtp);
+	ast_rtp_glue_unregister(&gtalk_rtp_glue);
 
 	if (!ast_mutex_lock(&gtalklock)) {
 		/* Hangup all interfaces if they have an owner */

channels/chan_h323.c

@@ -76,7 +76,7 @@ extern "C" {
 #include "asterisk/utils.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/cli.h"
@@ -161,7 +161,7 @@ struct oh323_pvt {
 	char accountcode[256];			/*!< Account code */
 	char rdnis[80];				/*!< Referring DNIS, if available */
 	int amaflags;				/*!< AMA Flags */
-	struct ast_rtp *rtp;			/*!< RTP Session */
+	struct ast_rtp_instance *rtp;		/*!< RTP Session */
 	struct ast_dsp *vad;			/*!< Used for in-band DTMF detection */
 	int nativeformats;			/*!< Codec formats supported by a channel */
 	int needhangup;				/*!< Send hangup when Asterisk is ready */
@@ -254,7 +254,7 @@ static const struct ast_channel_tech oh323_tech = {
 	.write = oh323_write,
 	.indicate = oh323_indicate,
 	.fixup = oh323_fixup,
-	.bridge = ast_rtp_bridge,
+	.bridge = ast_rtp_instance_bridge,
 };
 
 static const char* redirectingreason2str(int redirectingreason)
@@ -381,8 +381,8 @@ static void __oh323_update_info(struct ast_channel *c, struct oh323_pvt *pvt)
 	if (pvt->update_rtp_info > 0) {
 		if (pvt->rtp) {
 			ast_jb_configure(c, &global_jbconf);
-			ast_channel_set_fd(c, 0, ast_rtp_fd(pvt->rtp));
-			ast_channel_set_fd(c, 1, ast_rtcp_fd(pvt->rtp));
+			ast_channel_set_fd(c, 0, ast_rtp_instance_fd(pvt->rtp, 0));
+			ast_channel_set_fd(c, 1, ast_rtp_instance_fd(pvt->rtp, 1));
 			ast_queue_frame(pvt->owner, &ast_null_frame);	/* Tell Asterisk to apply changes */
 		}
 		pvt->update_rtp_info = -1;
@@ -444,7 +444,7 @@ static void __oh323_destroy(struct oh323_pvt *pvt)
 	AST_SCHED_DEL(sched, pvt->DTMFsched);
 
 	if (pvt->rtp) {
-		ast_rtp_destroy(pvt->rtp);
+		ast_rtp_instance_destroy(pvt->rtp);
 	}
 
 	/* Free dsp used for in-band DTMF detection */
@@ -510,7 +510,7 @@ static int oh323_digit_begin(struct ast_channel *c, char digit)
 		if (h323debug) {
 			ast_log(LOG_DTMF, "Begin sending out-of-band digit %c on %s\n", digit, c->name);
 		}
-		ast_rtp_senddigit_begin(pvt->rtp, digit);
+		ast_rtp_instance_dtmf_begin(pvt->rtp, digit);
 		ast_mutex_unlock(&pvt->lock);
 	} else if (pvt->txDtmfDigit != digit) {
 		/* in-band DTMF */
@@ -549,7 +549,7 @@ static int oh323_digit_end(struct ast_channel *c, char digit, unsigned int durat
 		if (h323debug) {
 			ast_log(LOG_DTMF, "End sending out-of-band digit %c on %s, duration %d\n", digit, c->name, duration);
 		}
-		ast_rtp_senddigit_end(pvt->rtp, digit);
+		ast_rtp_instance_dtmf_end(pvt->rtp, digit);
 		ast_mutex_unlock(&pvt->lock);
 	} else {
 		/* in-band DTMF */
@@ -747,11 +747,11 @@ static struct ast_frame *oh323_rtp_read(struct oh323_pvt *pvt)
 
 	/* Only apply it for the first packet, we just need the correct ip/port */
 	if (pvt->options.nat) {
-		ast_rtp_setnat(pvt->rtp, pvt->options.nat);
+		ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
 		pvt->options.nat = 0;
 	}
 
-	f = ast_rtp_read(pvt->rtp);
+	f = ast_rtp_instance_read(pvt->rtp, 0);
 	/* Don't send RFC2833 if we're not supposed to */
 	if (f && (f->frametype == AST_FRAME_DTMF) && !(pvt->options.dtmfmode & (H323_DTMF_RFC2833 | H323_DTMF_CISCO))) {
 		return &ast_null_frame;
@@ -808,7 +808,7 @@ static struct ast_frame *oh323_read(struct ast_channel *c)
 		break;
 	case 1:
 		if (pvt->rtp)
-			fr = ast_rtcp_read(pvt->rtp);
+			fr = ast_rtp_instance_read(pvt->rtp, 1);
 		else
 			fr = &ast_null_frame;
 		break;
@@ -842,7 +842,7 @@ static int oh323_write(struct ast_channel *c, struct ast_frame *frame)
 	if (pvt) {
 		ast_mutex_lock(&pvt->lock);
 		if (pvt->rtp && !pvt->recvonly)
-			res = ast_rtp_write(pvt->rtp, frame);
+			res = ast_rtp_instance_write(pvt->rtp, frame);
 		__oh323_update_info(c, pvt);
 		ast_mutex_unlock(&pvt->lock);
 	}
@@ -910,7 +910,7 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
 		res = 0;
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(pvt->rtp);
+		ast_rtp_instance_new_source(pvt->rtp);
 		res = 0;
 		break;
 	case AST_CONTROL_PROCEEDING:
@@ -946,17 +946,17 @@ static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
 
 static int __oh323_rtp_create(struct oh323_pvt *pvt)
 {
-	struct in_addr our_addr;
+	struct sockaddr_in our_addr;
 
 	if (pvt->rtp)
 		return 0;
 
-	if (ast_find_ourip(&our_addr, bindaddr)) {
+	if (ast_find_ourip(&our_addr.sin_addr, bindaddr)) {
 		ast_mutex_unlock(&pvt->lock);
 		ast_log(LOG_ERROR, "Unable to locate local IP address for RTP stream\n");
 		return -1;
 	}
-	pvt->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, our_addr);
+	pvt->rtp = ast_rtp_instance_new(NULL, sched, &our_addr, NULL);
 	if (!pvt->rtp) {
 		ast_mutex_unlock(&pvt->lock);
 		ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
@@ -965,24 +965,24 @@ static int __oh323_rtp_create(struct oh323_pvt *pvt)
 	if (h323debug)
 		ast_debug(1, "Created RTP channel\n");
 
-	ast_rtp_setqos(pvt->rtp, tos, cos, "H323 RTP");
+	ast_rtp_instance_set_qos(pvt->rtp, tos, cos, "H323 RTP");
 
 	if (h323debug)
 		ast_debug(1, "Setting NAT on RTP to %d\n", pvt->options.nat);
-	ast_rtp_setnat(pvt->rtp, pvt->options.nat);
+	ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
 
 	if (pvt->dtmf_pt[0] > 0)
-		ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
 	if (pvt->dtmf_pt[1] > 0)
-		ast_rtp_set_rtpmap_type(pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
 
 	if (pvt->peercapability)
-		ast_rtp_codec_setpref(pvt->rtp, &pvt->peer_prefs);
+		ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
 
 	if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
 		ast_jb_configure(pvt->owner, &global_jbconf);
-		ast_channel_set_fd(pvt->owner, 0, ast_rtp_fd(pvt->rtp));
-		ast_channel_set_fd(pvt->owner, 1, ast_rtcp_fd(pvt->rtp));
+		ast_channel_set_fd(pvt->owner, 0, ast_rtp_instance_fd(pvt->rtp, 0));
+		ast_channel_set_fd(pvt->owner, 1, ast_rtp_instance_fd(pvt->rtp, 1));
 		ast_queue_frame(pvt->owner, &ast_null_frame);	/* Tell Asterisk to apply changes */
 		ast_channel_unlock(pvt->owner);
 	} else
@@ -1028,13 +1028,13 @@ static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const c
 		if (!pvt->rtp)
 			__oh323_rtp_create(pvt);
 #if 0
-		ast_channel_set_fd(ch, 0, ast_rtp_fd(pvt->rtp));
-		ast_channel_set_fd(ch, 1, ast_rtcp_fd(pvt->rtp));
+		ast_channel_set_fd(ch, 0, ast_rtp_instance_fd(pvt->rtp, 0));
+		ast_channel_set_fd(ch, 1, ast_rtp_instance_fd(pvt->rtp, 1));
 #endif
 #ifdef VIDEO_SUPPORT
 		if (pvt->vrtp) {
-			ast_channel_set_fd(ch, 2, ast_rtp_fd(pvt->vrtp));
-			ast_channel_set_fd(ch, 3, ast_rtcp_fd(pvt->vrtp));
+			ast_channel_set_fd(ch, 2, ast_rtp_instance_fd(pvt->vrtp, 0));
+			ast_channel_set_fd(ch, 3, ast_rtp_instance_fd(pvt->vrtp, 1));
 		}
 #endif
 #ifdef T38_SUPPORT
@@ -1112,7 +1112,7 @@ static struct oh323_pvt *oh323_alloc(int callid)
 		}
 		if (!pvt->cd.call_token) {
 			ast_log(LOG_ERROR, "Not enough memory to alocate call token\n");
-			ast_rtp_destroy(pvt->rtp);
+			ast_rtp_instance_destroy(pvt->rtp);
 			ast_free(pvt);
 			return NULL;
 		}
@@ -1912,7 +1912,7 @@ static struct rtp_info *external_rtp_create(unsigned call_reference, const char
 		return NULL;
 	}
 	/* figure out our local RTP port and tell the H.323 stack about it */
-	ast_rtp_get_us(pvt->rtp, &us);
+	ast_rtp_instance_get_local_address(pvt->rtp, &us);
 	ast_mutex_unlock(&pvt->lock);
 
 	ast_copy_string(info->addr, ast_inet_ntoa(us.sin_addr), sizeof(info->addr));
@@ -1931,7 +1931,6 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
 {
 	struct oh323_pvt *pvt;
 	struct sockaddr_in them;
-	struct rtpPayloadType rtptype;
 	int nativeformats_changed;
 	enum { NEED_NONE, NEED_HOLD, NEED_UNHOLD } rtp_change = NEED_NONE;
 
@@ -1953,7 +1952,7 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
 		__oh323_rtp_create(pvt);
 
 	if ((pt == 2) && (pvt->jointcapability & AST_FORMAT_G726_AAL2)) {
-		ast_rtp_set_rtpmap_type(pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
 	}
 
 	them.sin_family = AF_INET;
@@ -1962,13 +1961,13 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
 	them.sin_port = htons(remotePort);
 
 	if (them.sin_addr.s_addr) {
-		ast_rtp_set_peer(pvt->rtp, &them);
+		ast_rtp_instance_set_remote_address(pvt->rtp, &them);
 		if (pvt->recvonly) {
 			pvt->recvonly = 0;
 			rtp_change = NEED_UNHOLD;
 		}
 	} else {
-		ast_rtp_stop(pvt->rtp);
+		ast_rtp_instance_stop(pvt->rtp);
 		if (!pvt->recvonly) {
 			pvt->recvonly = 1;
 			rtp_change = NEED_HOLD;
@@ -1978,7 +1977,7 @@ static void setup_rtp_connection(unsigned call_reference, const char *remoteIp,
 	/* Change native format to reflect information taken from OLC/OLCAck */
 	nativeformats_changed = 0;
 	if (pt != 128 && pvt->rtp) {	/* Payload type is invalid, so try to use previously decided */
-		rtptype = ast_rtp_lookup_pt(pvt->rtp, pt);
+		struct ast_rtp_payload_type rtptype = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(pvt->rtp), pt);
 		if (h323debug)
 			ast_debug(1, "Native format is set to %d from %d by RTP payload type %d\n", rtptype.code, pvt->nativeformats, pt);
 		if (pvt->nativeformats != rtptype.code) {
@@ -2359,7 +2358,7 @@ static void cleanup_connection(unsigned call_reference, const char *call_token)
 	}
 	if (pvt->rtp) {
 		/* Immediately stop RTP */
-		ast_rtp_destroy(pvt->rtp);
+		ast_rtp_instance_destroy(pvt->rtp);
 		pvt->rtp = NULL;
 	}
 	/* Free dsp used for in-band DTMF detection */
@@ -2421,7 +2420,7 @@ static void set_dtmf_payload(unsigned call_reference, const char *token, int pay
 		return;
 	}
 	if (pvt->rtp) {
-		ast_rtp_set_rtpmap_type(pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
 	}
 	pvt->dtmf_pt[is_cisco ? 1 : 0] = payload;
 	ast_mutex_unlock(&pvt->lock);
@@ -2452,7 +2451,7 @@ static void set_peer_capabilities(unsigned call_reference, const char *token, in
 			}
 		}
 		if (pvt->rtp)
-			ast_rtp_codec_setpref(pvt->rtp, &pvt->peer_prefs);
+			ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
 	}
 	ast_mutex_unlock(&pvt->lock);
 }
@@ -3113,19 +3112,19 @@ static int reload(void)
 static struct ast_cli_entry cli_h323_reload =
 	AST_CLI_DEFINE(handle_cli_h323_reload, "Reload H.323 configuration");
 
-static enum ast_rtp_get_result oh323_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result oh323_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
 	struct oh323_pvt *pvt;
-	enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+	enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
 
 	if (!(pvt = (struct oh323_pvt *)chan->tech_pvt))
-		return AST_RTP_GET_FAILED;
+		return AST_RTP_GLUE_RESULT_FORBID;
 
 	ast_mutex_lock(&pvt->lock);
-	*rtp = pvt->rtp;
+	*instance = pvt->rtp ? ao2_ref(pvt->rtp, +1), pvt->rtp : NULL;
 #if 0
 	if (pvt->options.bridge) {
-		res = AST_RTP_TRY_NATIVE;
+		res = AST_RTP_GLUE_RESULT_REMOTE;
 	}
 #endif
 	ast_mutex_unlock(&pvt->lock);
@@ -3133,11 +3132,6 @@ static enum ast_rtp_get_result oh323_get_rtp_peer(struct ast_channel *chan, stru
 	return res;
 }
 
-static enum ast_rtp_get_result oh323_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
-{
-	return AST_RTP_GET_FAILED;
-}
-
 static char *convertcap(int cap)
 {
 	switch (cap) {
@@ -3165,7 +3159,7 @@ static char *convertcap(int cap)
 	}
 }
 
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
 {
 	/* XXX Deal with Video */
 	struct oh323_pvt *pvt;
@@ -3183,19 +3177,18 @@ static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, str
 		ast_log(LOG_ERROR, "No Private Structure, this is bad\n");
 		return -1;
 	}
-	ast_rtp_get_peer(rtp, &them);
-	ast_rtp_get_us(rtp, &us);
+	ast_rtp_instance_get_remote_address(rtp, &them);
+	ast_rtp_instance_get_local_address(rtp, &us);
 #if 0	/* Native bridge still isn't ready */
 	h323_native_bridge(pvt->cd.call_token, ast_inet_ntoa(them.sin_addr), mode);
 #endif
 	return 0;
 }
 
-static struct ast_rtp_protocol oh323_rtp = {
+static struct ast_rtp_glue oh323_rtp_glue = {
 	.type = "H323",
 	.get_rtp_info = oh323_get_rtp_peer,
-	.get_vrtp_info = oh323_get_vrtp_peer,
-	.set_rtp_peer = oh323_set_rtp_peer,
+	.update_peer = oh323_set_rtp_peer,
 };
 
 static enum ast_module_load_result load_module(void)
@@ -3250,7 +3243,7 @@ static enum ast_module_load_result load_module(void)
 		}
 		ast_cli_register_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
 
-		ast_rtp_proto_register(&oh323_rtp);
+		ast_rtp_glue_register(&oh323_rtp_glue);
 
 		/* Register our callback functions */
 		h323_callback_register(setup_incoming_call,
@@ -3271,7 +3264,7 @@ static enum ast_module_load_result load_module(void)
 		/* start the h.323 listener */
 		if (h323_start_listener(h323_signalling_port, bindaddr)) {
 			ast_log(LOG_ERROR, "Unable to create H323 listener.\n");
-			ast_rtp_proto_unregister(&oh323_rtp);
+			ast_rtp_glue_unregister(&oh323_rtp_glue);
 			ast_cli_unregister_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
 			ast_cli_unregister(&cli_h323_reload);
 			h323_end_process();
@@ -3310,7 +3303,7 @@ static int unload_module(void)
 	ast_cli_unregister(&cli_h323_reload);
 
 	ast_channel_unregister(&oh323_tech);
-	ast_rtp_proto_unregister(&oh323_rtp);
+	ast_rtp_glue_unregister(&oh323_rtp_glue);
 
 	if (!ast_mutex_lock(&iflock)) {
 		/* hangup all interfaces if they have an owner */

channels/chan_jingle.c

@@ -53,7 +53,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/file.h"
@@ -112,9 +112,9 @@ struct jingle_pvt {
 	char exten[80];                  /*!< Called extension */
 	struct ast_channel *owner;       /*!< Master Channel */
 	char audio_content_name[100];    /*!< name attribute of content tag */
-	struct ast_rtp *rtp;             /*!< RTP audio session */
+	struct ast_rtp_instance *rtp;             /*!< RTP audio session */
 	char video_content_name[100];    /*!< name attribute of content tag */
-	struct ast_rtp *vrtp;            /*!< RTP video session */
+	struct ast_rtp_instance *vrtp;            /*!< RTP video session */
 	int jointcapability;             /*!< Supported capability at both ends (codecs ) */
 	int peercapability;
 	struct jingle_pvt *next;	/* Next entity */
@@ -183,11 +183,6 @@ static int jingle_sendhtml(struct ast_channel *ast, int subclass, const char *da
 static struct jingle_pvt *jingle_alloc(struct jingle *client, const char *from, const char *sid);
 static char *jingle_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
 static char *jingle_do_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-/*----- RTP interface functions */
-static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
-							   struct ast_rtp *vrtp, struct ast_rtp *tpeer, int codecs, int nat_active);
-static enum ast_rtp_get_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static int jingle_get_codec(struct ast_channel *chan);
 
 /*! \brief PBX interface structure for channel registration */
 static const struct ast_channel_tech jingle_tech = {
@@ -197,7 +192,7 @@ static const struct ast_channel_tech jingle_tech = {
 	.requester = jingle_request,
 	.send_digit_begin = jingle_digit_begin,
 	.send_digit_end = jingle_digit_end,
-	.bridge = ast_rtp_bridge,
+	.bridge = ast_rtp_instance_bridge,
 	.call = jingle_call,
 	.hangup = jingle_hangup,
 	.answer = jingle_answer,
@@ -216,15 +211,6 @@ static struct sched_context *sched;	/*!< The scheduling context */
 static struct io_context *io;	/*!< The IO context */
 static struct in_addr __ourip;
 
-
-/*! \brief RTP driver interface */
-static struct ast_rtp_protocol jingle_rtp = {
-	type: "Jingle",
-	get_rtp_info: jingle_get_rtp_peer,
-	set_rtp_peer: jingle_set_rtp_peer,
-	get_codec: jingle_get_codec,
-};
-
 static struct ast_cli_entry jingle_cli[] = {
 	AST_CLI_DEFINE(jingle_do_reload, "Reload Jingle configuration"),
 	AST_CLI_DEFINE(jingle_show_channels, "Show Jingle channels"),
@@ -304,7 +290,6 @@ static void add_codec_to_answer(const struct jingle_pvt *p, int codec, iks *dcod
 		iks_insert_attrib(payload_g723, "name", "G723");
 		iks_insert_node(dcodecs, payload_g723);
 	}
-	ast_rtp_lookup_code(p->rtp, 1, codec);
 }
 
 static int jingle_accept_call(struct jingle *client, struct jingle_pvt *p)
@@ -398,18 +383,19 @@ static int jingle_answer(struct ast_channel *ast)
 	return res;
 }
 
-static enum ast_rtp_get_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
 	struct jingle_pvt *p = chan->tech_pvt;
-	enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
+	enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
 
 	if (!p)
 		return res;
 
 	ast_mutex_lock(&p->lock);
 	if (p->rtp) {
-		*rtp = p->rtp;
-		res = AST_RTP_TRY_PARTIAL;
+		ao2_ref(p->rtp, +1);
+		*instance = p->rtp;
+		res = AST_RTP_GLUE_RESULT_LOCAL;
 	}
 	ast_mutex_unlock(&p->lock);
 
@@ -422,7 +408,7 @@ static int jingle_get_codec(struct ast_channel *chan)
 	return p->peercapability;
 }
 
-static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *tpeer, int codecs, int nat_active)
+static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, int codecs, int nat_active)
 {
 	struct jingle_pvt *p;
 
@@ -442,6 +428,13 @@ static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, st
 	return 0;
 }
 
+static struct ast_rtp_glue jingle_rtp_glue = {
+	.type = "Jingle",
+	.get_rtp_info = jingle_get_rtp_peer,
+	.get_codec = jingle_get_codec,
+	.update_peer = jingle_set_rtp_peer,
+};
+
 static int jingle_response(struct jingle *client, ikspak *pak, const char *reasonstr, const char *reasonstr2)
 {
 	iks *response = NULL, *error = NULL, *reason = NULL;
@@ -621,7 +614,7 @@ static int jingle_create_candidates(struct jingle *client, struct jingle_pvt *p,
 		goto safeout;
 	}
 
-	ast_rtp_get_us(p->rtp, &sin);
+	ast_rtp_instance_get_local_address(p->rtp, &sin);
 	ast_find_ourip(&us, bindaddr);
 
 	/* Setup our first jingle candidate */
@@ -779,7 +772,7 @@ static struct jingle_pvt *jingle_alloc(struct jingle *client, const char *from,
 		ast_copy_string(tmp->them, idroster, sizeof(tmp->them));
 		tmp->initiator = 1;
 	}
-	tmp->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+	tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
 	tmp->parent = client;
 	if (!tmp->rtp) {
 		ast_log(LOG_WARNING, "Out of RTP sessions?\n");
@@ -825,18 +818,18 @@ static struct ast_channel *jingle_new(struct jingle *client, struct jingle_pvt *
 
 	/* Set Frame packetization */
 	if (i->rtp)
-		ast_rtp_codec_setpref(i->rtp, &i->prefs);
+		ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
 
 	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
 	fmt = ast_best_codec(tmp->nativeformats);
 
 	if (i->rtp) {
-		ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
-		ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
+		ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
+		ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
 	}
 	if (i->vrtp) {
-		ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
-		ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
+		ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
+		ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
 	}
 	if (state == AST_STATE_RING)
 		tmp->rings = 1;
@@ -942,9 +935,9 @@ static void jingle_free_pvt(struct jingle *client, struct jingle_pvt *p)
 	if (p->owner)
 		ast_log(LOG_WARNING, "Uh oh, there's an owner, this is going to be messy.\n");
 	if (p->rtp)
-		ast_rtp_destroy(p->rtp);
+		ast_rtp_instance_destroy(p->rtp);
 	if (p->vrtp)
-		ast_rtp_destroy(p->vrtp);
+		ast_rtp_instance_destroy(p->vrtp);
 	jingle_free_candidates(p->theircandidates);
 	ast_free(p);
 }
@@ -1009,8 +1002,8 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
 		        ast_copy_string(p->audio_content_name, iks_find_attrib(content, "name"), sizeof(p->audio_content_name));
 
 			while (codec) {
-				ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
-				ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+				ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+				ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
 				codec = iks_next(codec);
 			}
 		}
@@ -1025,8 +1018,8 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
 		        ast_copy_string(p->video_content_name, iks_find_attrib(content, "name"), sizeof(p->video_content_name));
 
 			while (codec) {
-				ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
-				ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+				ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+				ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
 				codec = iks_next(codec);
 			}
 		}
@@ -1079,7 +1072,7 @@ static int jingle_update_stun(struct jingle *client, struct jingle_pvt *p)
 		sin.sin_port = htons(tmp->port);
 		snprintf(username, sizeof(username), "%s:%s", tmp->ufrag, p->ourcandidates->ufrag);
 
-		ast_rtp_stun_request(p->rtp, &sin, username);
+		ast_rtp_instance_stun_request(p->rtp, &sin, username);
 		tmp = tmp->next;
 	}
 	return 1;
@@ -1169,7 +1162,7 @@ static struct ast_frame *jingle_rtp_read(struct ast_channel *ast, struct jingle_
 
 	if (!p->rtp)
 		return &ast_null_frame;
-	f = ast_rtp_read(p->rtp);
+	f = ast_rtp_instance_read(p->rtp, 0);
 	jingle_update_stun(p->parent, p);
 	if (p->owner) {
 		/* We already hold the channel lock */
@@ -1220,7 +1213,7 @@ static int jingle_write(struct ast_channel *ast, struct ast_frame *frame)
 		if (p) {
 			ast_mutex_lock(&p->lock);
 			if (p->rtp) {
-				res = ast_rtp_write(p->rtp, frame);
+				res = ast_rtp_instance_write(p->rtp, frame);
 			}
 			ast_mutex_unlock(&p->lock);
 		}
@@ -1229,7 +1222,7 @@ static int jingle_write(struct ast_channel *ast, struct ast_frame *frame)
 		if (p) {
 			ast_mutex_lock(&p->lock);
 			if (p->vrtp) {
-				res = ast_rtp_write(p->vrtp, frame);
+				res = ast_rtp_instance_write(p->vrtp, frame);
 			}
 			ast_mutex_unlock(&p->lock);
 		}
@@ -1879,7 +1872,7 @@ static int load_module(void)
 		return 0;
 	}
 
-	ast_rtp_proto_register(&jingle_rtp);
+	ast_rtp_glue_register(&jingle_rtp_glue);
 	ast_cli_register_multiple(jingle_cli, ARRAY_LEN(jingle_cli));
 	/* Make sure we can register our channel type */
 	if (ast_channel_register(&jingle_tech)) {
@@ -1902,7 +1895,7 @@ static int unload_module(void)
 	ast_cli_unregister_multiple(jingle_cli, ARRAY_LEN(jingle_cli));
 	/* First, take us out of the channel loop */
 	ast_channel_unregister(&jingle_tech);
-	ast_rtp_proto_unregister(&jingle_rtp);
+	ast_rtp_glue_unregister(&jingle_rtp_glue);
 
 	if (!ast_mutex_lock(&jinglelock)) {
 		/* Hangup all interfaces if they have an owner */

channels/chan_local.c

@@ -39,7 +39,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/file.h"

channels/chan_mgcp.c

@@ -52,7 +52,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
 #include "asterisk/cli.h"
@@ -282,7 +282,7 @@ struct mgcp_subchannel {
 	int id;
 	struct ast_channel *owner;
 	struct mgcp_endpoint *parent;
-	struct ast_rtp *rtp;
+	struct ast_rtp_instance *rtp;
 	struct sockaddr_in tmpdest;
 	char txident[80]; /*! \todo FIXME txident is replaced by rqnt_ident in endpoint. 
 			This should be obsoleted */
@@ -408,7 +408,7 @@ static int transmit_response(struct mgcp_subchannel *sub, char *msg, struct mgcp
 static int transmit_notify_request(struct mgcp_subchannel *sub, char *tone);
 static int transmit_modify_request(struct mgcp_subchannel *sub);
 static int transmit_notify_request_with_callerid(struct mgcp_subchannel *sub, char *tone, char *callernum, char *callername);
-static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs);
+static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp, int codecs);
 static int transmit_connection_del(struct mgcp_subchannel *sub);
 static int transmit_audit_endpoint(struct mgcp_endpoint *p);
 static void start_rtp(struct mgcp_subchannel *sub);
@@ -447,7 +447,7 @@ static const struct ast_channel_tech mgcp_tech = {
 	.fixup = mgcp_fixup,
 	.send_digit_begin = mgcp_senddigit_begin,
 	.send_digit_end = mgcp_senddigit_end,
-	.bridge = ast_rtp_bridge,
+	.bridge = ast_rtp_instance_bridge,
 };
 
 static void mwi_event_cb(const struct ast_event *event, void *userdata)
@@ -503,7 +503,7 @@ static int unalloc_sub(struct mgcp_subchannel *sub)
 	sub->alreadygone = 0;
 	memset(&sub->tmpdest, 0, sizeof(sub->tmpdest));
 	if (sub->rtp) {
-		ast_rtp_destroy(sub->rtp);
+		ast_rtp_instance_destroy(sub->rtp);
 		sub->rtp = NULL;
 	}
 	dump_cmd_queues(NULL, sub); /* SC */
@@ -1003,7 +1003,7 @@ static int mgcp_hangup(struct ast_channel *ast)
 	/* Reset temporary destination */
 	memset(&sub->tmpdest, 0, sizeof(sub->tmpdest));
 	if (sub->rtp) {
-		ast_rtp_destroy(sub->rtp);
+		ast_rtp_instance_destroy(sub->rtp);
 		sub->rtp = NULL;
 	}
 
@@ -1203,7 +1203,7 @@ static struct ast_frame *mgcp_rtp_read(struct mgcp_subchannel *sub)
 	/* Retrieve audio/etc from channel.  Assumes sub->lock is already held. */
 	struct ast_frame *f;
 
-	f = ast_rtp_read(sub->rtp);
+	f = ast_rtp_instance_read(sub->rtp, 0);
 	/* Don't send RFC2833 if we're not supposed to */
 	if (f && (f->frametype == AST_FRAME_DTMF) && !(sub->parent->dtmfmode & MGCP_DTMF_RFC2833))
 		return &ast_null_frame;
@@ -1261,7 +1261,7 @@ static int mgcp_write(struct ast_channel *ast, struct ast_frame *frame)
 		ast_mutex_lock(&sub->lock);
 		if ((sub->parent->sub == sub) || !sub->parent->singlepath) {
 			if (sub->rtp) {
-				res =  ast_rtp_write(sub->rtp, frame);
+				res =  ast_rtp_instance_write(sub->rtp, frame);
 			}
 		}
 		ast_mutex_unlock(&sub->lock);
@@ -1297,7 +1297,7 @@ static int mgcp_senddigit_begin(struct ast_channel *ast, char digit)
 		res = -1; /* Let asterisk play inband indications */
 	} else if (p->dtmfmode & MGCP_DTMF_RFC2833) {
 		ast_log(LOG_DEBUG, "Sending DTMF using RFC2833");
-		ast_rtp_senddigit_begin(sub->rtp, digit);
+		ast_rtp_instance_dtmf_begin(sub->rtp, digit);
 	} else {
 		ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
 	}
@@ -1324,7 +1324,7 @@ static int mgcp_senddigit_end(struct ast_channel *ast, char digit, unsigned int
 		tmp[2] = digit;
 		tmp[3] = '\0';
 		transmit_notify_request(sub, tmp);
-                ast_rtp_senddigit_end(sub->rtp, digit);
+                ast_rtp_instance_dtmf_end(sub->rtp, digit);
 	} else {
 		ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
 	}
@@ -1453,7 +1453,7 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz
 		ast_moh_stop(ast);
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(sub->rtp);
+		ast_rtp_instance_new_source(sub->rtp);
 		break;
 	case -1:
 		transmit_notify_request(sub, "");
@@ -1481,7 +1481,7 @@ static struct ast_channel *mgcp_new(struct mgcp_subchannel *sub, int state)
 		fmt = ast_best_codec(tmp->nativeformats);
 		ast_string_field_build(tmp, name, "MGCP/%s@%s-%d", i->name, i->parent->name, sub->id);
 		if (sub->rtp)
-			ast_channel_set_fd(tmp, 0, ast_rtp_fd(sub->rtp));
+			ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(sub->rtp, 0));
 		if (i->dtmfmode & (MGCP_DTMF_INBAND | MGCP_DTMF_HYBRID)) {
 			i->dsp = ast_dsp_new();
 			ast_dsp_set_features(i->dsp, DSP_FEATURE_DIGIT_DETECT);
@@ -1874,12 +1874,12 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
 	sin.sin_family = AF_INET;
 	memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
 	sin.sin_port = htons(portno);
-	ast_rtp_set_peer(sub->rtp, &sin);
+	ast_rtp_instance_set_remote_address(sub->rtp, &sin);
 #if 0
 	printf("Peer RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
 #endif	
 	/* Scan through the RTP payload types specified in a "m=" line: */
-	ast_rtp_pt_clear(sub->rtp);
+	ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp);
 	codecs = ast_strdupa(m + len);
 	while (!ast_strlen_zero(codecs)) {
 		if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
@@ -1888,7 +1888,7 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
 			ast_log(LOG_WARNING, "Error in codec string '%s' at '%s'\n", m, codecs);
 			return -1;
 		}
-		ast_rtp_set_m_type(sub->rtp, codec);
+		ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, codec);
 		codec_count++;
 		codecs += len;
 	}
@@ -1901,11 +1901,11 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
 		if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
 			continue;
 		/* Note: should really look at the 'freq' and '#chans' params too */
-		ast_rtp_set_rtpmap_type(sub->rtp, codec, "audio", mimeSubtype, 0);
+		ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, codec, "audio", mimeSubtype, 0);
 	}
 
 	/* Now gather all of the codecs that were asked for: */
-	ast_rtp_get_current_formats(sub->rtp, &peercapability, &peerNonCodecCapability);
+	ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(sub->rtp), &peercapability, &peerNonCodecCapability);
 	p->capability = capability & peercapability;
 	if (mgcpdebug) {
 		ast_verbose("Capabilities: us - %d, them - %d, combined - %d\n",
@@ -2043,7 +2043,7 @@ static int transmit_response(struct mgcp_subchannel *sub, char *msg, struct mgcp
 }
 
 
-static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struct ast_rtp *rtp)
+static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp)
 {
 	int len;
 	int codec;
@@ -2066,9 +2066,9 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 		ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
 		return -1;
 	}
-	ast_rtp_get_us(sub->rtp, &sin);
+	ast_rtp_instance_get_local_address(sub->rtp, &sin);
 	if (rtp) {
-		ast_rtp_get_peer(rtp, &dest);
+		ast_rtp_instance_get_remote_address(sub->rtp, &dest);
 	} else {
 		if (sub->tmpdest.sin_addr.s_addr) {
 			dest.sin_addr = sub->tmpdest.sin_addr;
@@ -2094,11 +2094,11 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 			if (mgcpdebug) {
 				ast_verbose("Answering with capability %d\n", x);
 			}
-			codec = ast_rtp_lookup_code(sub->rtp, 1, x);
+			codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 1, x);
 			if (codec > -1) {
 				snprintf(costr, sizeof(costr), " %d", codec);
 				strncat(m, costr, sizeof(m) - strlen(m) - 1);
-				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x, 0));
+				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype2(1, x, 0));
 				strncat(a, costr, sizeof(a) - strlen(a) - 1);
 			}
 		}
@@ -2108,11 +2108,11 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 			if (mgcpdebug) {
 				ast_verbose("Answering with non-codec capability %d\n", x);
 			}
-			codec = ast_rtp_lookup_code(sub->rtp, 0, x);
+			codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 0, x);
 			if (codec > -1) {
 				snprintf(costr, sizeof(costr), " %d", codec);
 				strncat(m, costr, sizeof(m) - strlen(m) - 1);
-				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x, 0));
+				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype2(0, x, 0));
 				strncat(a, costr, sizeof(a) - strlen(a) - 1);
 				if (x == AST_RTP_DTMF) {
 					/* Indicate we support DTMF...  Not sure about 16,
@@ -2136,7 +2136,7 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 	return 0;
 }
 
-static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs)
+static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp, int codecs)
 {
 	struct mgcp_request resp;
 	char local[256];
@@ -2147,13 +2147,13 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	if (ast_strlen_zero(sub->cxident) && rtp) {
 		/* We don't have a CXident yet, store the destination and
 		   wait a bit */
-		ast_rtp_get_peer(rtp, &sub->tmpdest);
+		ast_rtp_instance_get_remote_address(rtp, &sub->tmpdest);
 		return 0;
 	}
 	ast_copy_string(local, "p:20", sizeof(local));
 	for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
 		if (p->capability & x) {
-			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
+			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype2(1, x, 0));
 			strncat(local, tmp, sizeof(local) - strlen(local) - 1);
 		}
 	}
@@ -2172,7 +2172,7 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	return send_request(p, sub, &resp, oseq); /* SC */
 }
 
-static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp)
+static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp)
 {
 	struct mgcp_request resp;
 	char local[256];
@@ -2183,7 +2183,7 @@ static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	ast_copy_string(local, "p:20", sizeof(local));
 	for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
 		if (p->capability & x) {
-			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
+			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype2(1, x, 0));
 			strncat(local, tmp, sizeof(local) - strlen(local) - 1);
 		}
 	}
@@ -2611,21 +2611,17 @@ static void start_rtp(struct mgcp_subchannel *sub)
 	ast_mutex_lock(&sub->lock);
 	/* check again to be on the safe side */
 	if (sub->rtp) {
-		ast_rtp_destroy(sub->rtp);
+		ast_rtp_instance_destroy(sub->rtp);
 		sub->rtp = NULL;
 	}
 	/* Allocate the RTP now */
-	sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+	sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
 	if (sub->rtp && sub->owner)
-		ast_channel_set_fd(sub->owner, 0, ast_rtp_fd(sub->rtp));
+		ast_channel_set_fd(sub->owner, 0, ast_rtp_instance_fd(sub->rtp, 0));
 	if (sub->rtp) {
-		ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "MGCP RTP");
-		ast_rtp_setnat(sub->rtp, sub->nat);
+		ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "MGCP RTP");
+		ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->nat);
 	}
-#if 0
-	ast_rtp_set_callback(p->rtp, rtpready);
-	ast_rtp_set_data(p->rtp, p);
-#endif		
 	/* Make a call*ID */
         snprintf(sub->callid, sizeof(sub->callid), "%08lx%s", ast_random(), sub->txident);
 	/* Transmit the connection create */
@@ -3940,22 +3936,22 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
 	return (gw_reload ? NULL : gw);
 }
 
-static enum ast_rtp_get_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
 	struct mgcp_subchannel *sub = NULL;
 
 	if (!(sub = chan->tech_pvt) || !(sub->rtp))
-		return AST_RTP_GET_FAILED;
+		return AST_RTP_GLUE_RESULT_FORBID;
 
-	*rtp = sub->rtp;
+	*instance = sub->rtp ? ao2_ref(sub->rtp, +1), sub->rtp : NULL;
 
 	if (sub->parent->canreinvite)
-		return AST_RTP_TRY_NATIVE;
+		return AST_RTP_GLUE_RESULT_REMOTE;
 	else
-		return AST_RTP_TRY_PARTIAL;
+		return AST_RTP_GLUE_RESULT_LOCAL;
 }
 
-static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
 {
 	/* XXX Is there such thing as video support with MGCP? XXX */
 	struct mgcp_subchannel *sub;
@@ -3967,10 +3963,10 @@ static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, stru
 	return -1;
 }
 
-static struct ast_rtp_protocol mgcp_rtp = {
+static struct ast_rtp_glue mgcp_rtp_glue = {
 	.type = "MGCP",
 	.get_rtp_info = mgcp_get_rtp_peer,
-	.set_rtp_peer = mgcp_set_rtp_peer,
+	.update_peer = mgcp_set_rtp_peer,
 };
 
 static void destroy_endpoint(struct mgcp_endpoint *e)
@@ -3984,7 +3980,7 @@ static void destroy_endpoint(struct mgcp_endpoint *e)
 			transmit_connection_del(sub);
 		}
 		if (sub->rtp) {
-			ast_rtp_destroy(sub->rtp);
+			ast_rtp_instance_destroy(sub->rtp);
 			sub->rtp = NULL;
 		}
 		memset(sub->magic, 0, sizeof(sub->magic));
@@ -4276,7 +4272,7 @@ static int load_module(void)
 		return AST_MODULE_LOAD_FAILURE;
 	}
 
-	ast_rtp_proto_register(&mgcp_rtp);
+	ast_rtp_glue_register(&mgcp_rtp_glue);
 	ast_cli_register_multiple(cli_mgcp, sizeof(cli_mgcp) / sizeof(struct ast_cli_entry));
 	
 	/* And start the monitor for the first time */
@@ -4379,7 +4375,7 @@ static int unload_module(void)
 	}
 
 	close(mgcpsock);
-	ast_rtp_proto_unregister(&mgcp_rtp);
+	ast_rtp_glue_unregister(&mgcp_rtp_glue);
 	ast_cli_unregister_multiple(cli_mgcp, sizeof(cli_mgcp) / sizeof(struct ast_cli_entry));
 	sched_context_destroy(sched);
 

channels/chan_skinny.c

@@ -49,7 +49,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/sched.h"
 #include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/netsock.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
@@ -1111,8 +1111,8 @@ static int matchdigittimeout = 3000;
 struct skinny_subchannel {
 	ast_mutex_t lock;
 	struct ast_channel *owner;
-	struct ast_rtp *rtp;
-	struct ast_rtp *vrtp;
+	struct ast_rtp_instance *rtp;
+	struct ast_rtp_instance *vrtp;
 	unsigned int callid;
 	/* time_t lastouttime; */ /* Unused */
 	int progress;
@@ -1347,7 +1347,7 @@ static const struct ast_channel_tech skinny_tech = {
 	.fixup = skinny_fixup,
 	.send_digit_begin = skinny_senddigit_begin,
 	.send_digit_end = skinny_senddigit_end,
-	.bridge = ast_rtp_bridge,  
+	.bridge = ast_rtp_instance_bridge, 
 };
 
 static int skinny_extensionstate_cb(char *context, char* exten, int state, void *data);
@@ -2557,46 +2557,48 @@ static void mwi_event_cb(const struct ast_event *event, void *userdata)
 /* I do not believe skinny can deal with video.
    Anyone know differently? */
 /* Yes, it can.  Currently 7985 and Cisco VT Advantage do video. */
-static enum ast_rtp_get_result skinny_get_vrtp_peer(struct ast_channel *c, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result skinny_get_vrtp_peer(struct ast_channel *c, struct ast_rtp_instance **instance)
 {
 	struct skinny_subchannel *sub = NULL;
 
 	if (!(sub = c->tech_pvt) || !(sub->vrtp))
-		return AST_RTP_GET_FAILED;
+		return AST_RTP_GLUE_RESULT_FORBID;
 
-	*rtp = sub->vrtp;
+	ao2_ref(sub->vrtp, +1);
+	*instance = sub->vrtp;
 
-	return AST_RTP_TRY_NATIVE;
+	return AST_RTP_GLUE_RESULT_REMOTE;
 }
 
-static enum ast_rtp_get_result skinny_get_rtp_peer(struct ast_channel *c, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result skinny_get_rtp_peer(struct ast_channel *c, struct ast_rtp_instance **instance)
 {
 	struct skinny_subchannel *sub = NULL;
 	struct skinny_line *l;
-	enum ast_rtp_get_result res = AST_RTP_TRY_NATIVE;
+	enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_REMOTE;
 
 	if (skinnydebug)
 		ast_verb(1, "skinny_get_rtp_peer() Channel = %s\n", c->name);
 
 
 	if (!(sub = c->tech_pvt))
-		return AST_RTP_GET_FAILED;
+		return AST_RTP_GLUE_RESULT_FORBID;
 
 	ast_mutex_lock(&sub->lock);
 
 	if (!(sub->rtp)){
 		ast_mutex_unlock(&sub->lock);
-		return AST_RTP_GET_FAILED;
+		return AST_RTP_GLUE_RESULT_FORBID;
 	}
-	
-	*rtp = sub->rtp;
+
+	ao2_ref(sub->rtp, +1);
+	*instance = sub->rtp;
 
 	l = sub->parent;
 
 	if (!l->canreinvite || l->nat){
-		res = AST_RTP_TRY_PARTIAL;
+		res = AST_RTP_GLUE_RESULT_LOCAL;
 		if (skinnydebug)
-			ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_TRY_PARTIAL \n");
+			ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_GLUE_RESULT_LOCAL \n");
 	}
 
 	ast_mutex_unlock(&sub->lock);
@@ -2605,7 +2607,7 @@ static enum ast_rtp_get_result skinny_get_rtp_peer(struct ast_channel *c, struct
 
 }
 
-static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
 {
 	struct skinny_subchannel *sub;
 	struct skinny_line *l;
@@ -2630,7 +2632,7 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
 	s = d->session;
 
 	if (rtp){
-		ast_rtp_get_peer(rtp, &them);
+		ast_rtp_instance_get_remote_address(rtp, &them);
 
 		/* Shutdown any early-media or previous media on re-invite */
 		if (!(req = req_alloc(sizeof(struct stop_media_transmission_message), STOP_MEDIA_TRANSMISSION_MESSAGE)))
@@ -2654,7 +2656,7 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
 		req->data.startmedia.conferenceId = htolel(sub->callid);
 		req->data.startmedia.passThruPartyId = htolel(sub->callid);
 		if (!(l->canreinvite) || (l->nat)){
-			ast_rtp_get_us(rtp, &us);
+			ast_rtp_instance_get_local_address(rtp, &us);
 			req->data.startmedia.remoteIp = htolel(d->ourip.s_addr);
 			req->data.startmedia.remotePort = htolel(ntohs(us.sin_port));
 		} else {
@@ -2675,11 +2677,11 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
 	return 0;
 }
 
-static struct ast_rtp_protocol skinny_rtp = {
+static struct ast_rtp_glue skinny_rtp_glue = {
 	.type = "Skinny",
 	.get_rtp_info = skinny_get_rtp_peer,
 	.get_vrtp_info = skinny_get_vrtp_peer,
-	.set_rtp_peer = skinny_set_rtp_peer,
+	.update_peer = skinny_set_rtp_peer,
 };
 
 static char *handle_skinny_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
@@ -3559,29 +3561,36 @@ static void start_rtp(struct skinny_subchannel *sub)
 
 	ast_mutex_lock(&sub->lock);
 	/* Allocate the RTP */
-	sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+	sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
 	if (hasvideo)
-		sub->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
-	
-	if (sub->rtp && sub->owner) {
-		ast_channel_set_fd(sub->owner, 0, ast_rtp_fd(sub->rtp));
-		ast_channel_set_fd(sub->owner, 1, ast_rtcp_fd(sub->rtp));
-	}
-	if (hasvideo && sub->vrtp && sub->owner) {
-		ast_channel_set_fd(sub->owner, 2, ast_rtp_fd(sub->vrtp));
-		ast_channel_set_fd(sub->owner, 3, ast_rtcp_fd(sub->vrtp));
-	}
+		sub->vrtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+
 	if (sub->rtp) {
-		ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "Skinny RTP");
-		ast_rtp_setnat(sub->rtp, l->nat);
+		ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
 	}
 	if (sub->vrtp) {
-		ast_rtp_setqos(sub->vrtp, qos.tos_video, qos.cos_video, "Skinny VRTP");
-		ast_rtp_setnat(sub->vrtp, l->nat);
+		ast_rtp_instance_set_prop(sub->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+	}
+
+	if (sub->rtp && sub->owner) {
+		ast_channel_set_fd(sub->owner, 0, ast_rtp_instance_fd(sub->rtp, 0));
+		ast_channel_set_fd(sub->owner, 1, ast_rtp_instance_fd(sub->rtp, 1));
+	}
+	if (hasvideo && sub->vrtp && sub->owner) {
+		ast_channel_set_fd(sub->owner, 2, ast_rtp_instance_fd(sub->vrtp, 0));
+		ast_channel_set_fd(sub->owner, 3, ast_rtp_instance_fd(sub->vrtp, 1));
+	}
+	if (sub->rtp) {
+		ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "Skinny RTP");
+		ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, l->nat);
+	}
+	if (sub->vrtp) {
+		ast_rtp_instance_set_qos(sub->vrtp, qos.tos_video, qos.cos_video, "Skinny VRTP");
+		ast_rtp_instance_set_prop(sub->vrtp, AST_RTP_PROPERTY_NAT, l->nat);
 	}
 	/* Set Frame packetization */
 	if (sub->rtp)
-		ast_rtp_codec_setpref(sub->rtp, &l->prefs);
+		ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, &l->prefs);
 
 	/* Create the RTP connection */
 	transmit_connect(d, sub);
@@ -3852,7 +3861,7 @@ static int skinny_hangup(struct ast_channel *ast)
 	sub->alreadygone = 0;
 	sub->outgoing = 0;
 	if (sub->rtp) {
-		ast_rtp_destroy(sub->rtp);
+		ast_rtp_instance_destroy(sub->rtp);
 		sub->rtp = NULL;
 	}
 	ast_mutex_unlock(&sub->lock);
@@ -3913,16 +3922,16 @@ static struct ast_frame *skinny_rtp_read(struct skinny_subchannel *sub)
 
 	switch(ast->fdno) {
 	case 0:
-		f = ast_rtp_read(sub->rtp); /* RTP Audio */
+		f = ast_rtp_instance_read(sub->rtp, 0); /* RTP Audio */
 		break;
 	case 1:
-		f = ast_rtcp_read(sub->rtp); /* RTCP Control Channel */
+		f = ast_rtp_instance_read(sub->rtp, 1); /* RTCP Control Channel */
 		break;
 	case 2:
-		f = ast_rtp_read(sub->vrtp); /* RTP Video */
+		f = ast_rtp_instance_read(sub->vrtp, 0); /* RTP Video */
 		break;
 	case 3:
-		f = ast_rtcp_read(sub->vrtp); /* RTCP Control Channel for video */
+		f = ast_rtp_instance_read(sub->vrtp, 1); /* RTCP Control Channel for video */
 		break;
 #if 0
 	case 5:
@@ -3979,7 +3988,7 @@ static int skinny_write(struct ast_channel *ast, struct ast_frame *frame)
 	if (sub) {
 		ast_mutex_lock(&sub->lock);
 		if (sub->rtp) {
-			res = ast_rtp_write(sub->rtp, frame);
+			res = ast_rtp_instance_write(sub->rtp, frame);
 		}
 		ast_mutex_unlock(&sub->lock);
 	}
@@ -4253,7 +4262,7 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
 	case AST_CONTROL_PROCEEDING:
 		break;
 	case AST_CONTROL_SRCUPDATE:
-		ast_rtp_new_source(sub->rtp);
+		ast_rtp_instance_new_source(sub->rtp);
 		break;
 	default:
 		ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
@@ -4312,7 +4321,7 @@ static struct ast_channel *skinny_new(struct skinny_line *l, int state)
 		if (skinnydebug)
 			ast_verb(1, "skinny_new: tmp->nativeformats=%d fmt=%d\n", tmp->nativeformats, fmt);
 		if (sub->rtp) {
-			ast_channel_set_fd(tmp, 0, ast_rtp_fd(sub->rtp));
+			ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(sub->rtp, 0));
 		}
 		if (state == AST_STATE_RING) {
 			tmp->rings = 1;
@@ -5537,8 +5546,8 @@ static int handle_open_receive_channel_ack_message(struct skinny_req *req, struc
 	l = sub->parent;
 
 	if (sub->rtp) {
-		ast_rtp_set_peer(sub->rtp, &sin);
-		ast_rtp_get_us(sub->rtp, &us);
+		ast_rtp_instance_set_remote_address(sub->rtp, &sin);
+		ast_rtp_instance_get_local_address(sub->rtp, &us);
 	} else {
 		ast_log(LOG_ERROR, "No RTP structure, this is very bad\n");
 		return 0;
@@ -7289,7 +7298,7 @@ static int load_module(void)
 		return -1;
 	}
 
-	ast_rtp_proto_register(&skinny_rtp);
+	ast_rtp_glue_register(&skinny_rtp_glue);
 	ast_cli_register_multiple(cli_skinny, ARRAY_LEN(cli_skinny));
 
 	ast_manager_register2("SKINNYdevices", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_skinny_show_devices,
@@ -7323,7 +7332,7 @@ static int unload_module(void)
 	struct skinny_subchannel *sub;
 	struct ast_context *con;
 
-	ast_rtp_proto_unregister(&skinny_rtp);
+	ast_rtp_glue_unregister(&skinny_rtp_glue);
 	ast_channel_unregister(&skinny_tech);
 	ast_cli_unregister_multiple(cli_skinny, ARRAY_LEN(cli_skinny));
 

channels/chan_unistim.c

@@ -60,7 +60,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/module.h"
 #include "asterisk/pbx.h"
 #include "asterisk/event.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
 #include "asterisk/netsock.h"
 #include "asterisk/acl.h"
 #include "asterisk/callerid.h"
@@ -365,7 +365,7 @@ struct unistim_subchannel {
 	/*! Unistim line */
 	struct unistim_line *parent;
 	/*! RTP handle */
-	struct ast_rtp *rtp;
+	struct ast_rtp_instance *rtp;
 	int alreadygone;
 	char ringvolume;
 	char ringstyle;
@@ -711,7 +711,7 @@ static const struct ast_channel_tech unistim_tech = {
 	.send_digit_begin = unistim_senddigit_begin,
 	.send_digit_end = unistim_senddigit_end,
 	.send_text = unistim_sendtext,
-/*      .bridge = ast_rtp_bridge, */
+	.bridge = ast_rtp_instance_bridge,
 };
 
 static void display_last_error(const char *sz_msg)
@@ -1854,7 +1854,7 @@ static void cancel_dial(struct unistimsession *pte)
 static void swap_subs(struct unistim_line *p, int a, int b)
 {
 /*  struct ast_channel *towner; */
-	struct ast_rtp *rtp;
+	struct ast_rtp_instance *rtp;
 	int fds;
 
 	if (unistimdebug)
@@ -2056,30 +2056,29 @@ static void start_rtp(struct unistim_subchannel *sub)
 	/* Allocate the RTP */
 	if (unistimdebug)
 		ast_verb(0, "Starting RTP. Bind on %s\n", ast_inet_ntoa(sout.sin_addr));
-	sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, sout.sin_addr);
+	sub->rtp = ast_rtp_instance_new(NULL, sched, &sout, NULL);
 	if (!sub->rtp) {
 		ast_log(LOG_WARNING, "Unable to create RTP session: %s binaddr=%s\n",
 				strerror(errno), ast_inet_ntoa(sout.sin_addr));
 		ast_mutex_unlock(&sub->lock);
 		return;
 	}
-	if (sub->rtp && sub->owner) {
-		sub->owner->fds[0] = ast_rtp_fd(sub->rtp);
-		sub->owner->fds[1] = ast_rtcp_fd(sub->rtp);
-	}
-	if (sub->rtp) {
-		ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
-		ast_rtp_setnat(sub->rtp, sub->parent->parent->nat);
+	ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
+	if (sub->owner) {
+		sub->owner->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
+		sub->owner->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
 	}
+	ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
+	ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->parent->parent->nat);
 
 	/* Create the RTP connection */
-	ast_rtp_get_us(sub->rtp, &us);
+	ast_rtp_instance_get_local_address(sub->rtp, &us);
 	sin.sin_family = AF_INET;
 	/* Setting up RTP for our side */
 	memcpy(&sin.sin_addr, &sub->parent->parent->session->sin.sin_addr,
 		   sizeof(sin.sin_addr));
 	sin.sin_port = htons(sub->parent->parent->rtp_port);
-	ast_rtp_set_peer(sub->rtp, &sin);
+	ast_rtp_instance_set_remote_address(sub->rtp, &sin);
 	if (!(sub->owner->nativeformats & sub->owner->readformat)) {
 		int fmt;
 		fmt = ast_best_codec(sub->owner->nativeformats);
@@ -2091,7 +2090,7 @@ static void start_rtp(struct unistim_subchannel *sub)
 		sub->owner->readformat = fmt;
 		sub->owner->writeformat = fmt;
 	}
-	codec = ast_rtp_lookup_code(sub->rtp, 1, sub->owner->readformat);
+	codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 1, sub->owner->readformat);
 	/* Setting up RTP of the phone */
 	if (public_ip.sin_family == 0)  /* NAT IP override ?   */
 		memcpy(&public, &us, sizeof(public));   /* No defined, using IP from recvmsg  */
@@ -3724,7 +3723,7 @@ static int unistim_hangup(struct ast_channel *ast)
 		if (sub->rtp) {
 			if (unistimdebug)
 				ast_verb(0, "Destroying RTP session\n");
-			ast_rtp_destroy(sub->rtp);
+			ast_rtp_instance_destroy(sub->rtp);
 			sub->rtp = NULL;
 		}
 		return 0;
@@ -3769,7 +3768,7 @@ static int unistim_hangup(struct ast_channel *ast)
 		if (sub->rtp) {
 			if (unistimdebug)
 				ast_verb(0, "Destroying RTP session\n");
-			ast_rtp_destroy(sub->rtp);
+			ast_rtp_instance_destroy(sub->rtp);
 			sub->rtp = NULL;
 		}
 		return 0;
@@ -3794,7 +3793,7 @@ static int unistim_hangup(struct ast_channel *ast)
 	if (sub->rtp) {
 		if (unistimdebug)
 			ast_verb(0, "Destroying RTP session\n");
-		ast_rtp_destroy(sub->rtp);
+		ast_rtp_instance_destroy(sub->rtp);
 		sub->rtp = NULL;
 	} else if (unistimdebug)
 		ast_verb(0, "No RTP session to destroy\n");
@@ -3921,10 +3920,10 @@ static struct ast_frame *unistim_rtp_read(const struct ast_channel *ast,
 
 	switch (ast->fdno) {
 	case 0:
-		f = ast_rtp_read(sub->rtp);     /* RTP Audio */
+		f = ast_rtp_instance_read(sub->rtp, 0);     /* RTP Audio */
 		break;
 	case 1:
-		f = ast_rtcp_read(sub->rtp);    /* RTCP Control Channel */
+		f = ast_rtp_instance_read(sub->rtp, 1);    /* RTCP Control Channel */
 		break;
 	default:
 		f = &ast_null_frame;
@@ -3990,7 +3989,7 @@ static int unistim_write(struct ast_channel *ast, struct ast_frame *frame)
 	if (sub) {
 		ast_mutex_lock(&sub->lock);
 		if (sub->rtp) {
-			res = ast_rtp_write(sub->rtp, frame);
+			res = ast_rtp_instance_write(sub->rtp, frame);
 		}
 		ast_mutex_unlock(&sub->lock);
 	}
@@ -4455,8 +4454,8 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
 	if ((sub->rtp) && (sub->subtype == 0)) {
 		if (unistimdebug)
 			ast_verb(0, "New unistim channel with a previous rtp handle ?\n");
-		tmp->fds[0] = ast_rtp_fd(sub->rtp);
-		tmp->fds[1] = ast_rtcp_fd(sub->rtp);
+		tmp->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
+		tmp->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
 	}
 	if (sub->rtp)
 		ast_jb_configure(tmp, &global_jbconf);
@@ -5526,51 +5525,19 @@ static int reload_config(void)
 	return 0;
 }
 
-static enum ast_rtp_get_result unistim_get_vrtp_peer(struct ast_channel *chan, 
-	struct ast_rtp **rtp)
+static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
-	return AST_RTP_TRY_NATIVE;
+	struct unistim_subchannel *sub = chan->tech_pvt;
+
+	ao2_ref(sub->rtp, +1);
+	*instance = sub->rtp;
+
+	return AST_RTP_GLUE_RESULT_LOCAL;
 }
 
-static enum ast_rtp_get_result unistim_get_rtp_peer(struct ast_channel *chan, 
-	struct ast_rtp **rtp)
-{
-	struct unistim_subchannel *sub;
-	enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
-
-	if (unistimdebug)
-		ast_verb(0, "unistim_get_rtp_peer called\n");
-		
-	sub = chan->tech_pvt;
-	if (sub && sub->rtp) {
-		*rtp = sub->rtp;
-		res = AST_RTP_TRY_NATIVE;
-	}
-
-	return res;
-}
-
-static int unistim_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, 
-	struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
-{
-	struct unistim_subchannel *sub;
-
-	if (unistimdebug)
-		ast_verb(0, "unistim_set_rtp_peer called\n");
-
-	sub = chan->tech_pvt;
-
-	if (sub)
-		return 0;
-
-	return -1;
-}
-
-static struct ast_rtp_protocol unistim_rtp = {
+static struct ast_rtp_glue unistim_rtp_glue = {
 	.type = channel_type,
 	.get_rtp_info = unistim_get_rtp_peer,
-	.get_vrtp_info = unistim_get_vrtp_peer,
-	.set_rtp_peer = unistim_set_rtp_peer,
 };
 
 /*--- load_module: PBX load module - initialization ---*/
@@ -5603,7 +5570,7 @@ int load_module(void)
 		goto chanreg_failed;
 	} 
 
-	ast_rtp_proto_register(&unistim_rtp);
+	ast_rtp_glue_register(&unistim_rtp_glue);
 
 	ast_cli_register_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
 
@@ -5634,7 +5601,7 @@ static int unload_module(void)
 	ast_cli_unregister_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
 
 	ast_channel_unregister(&unistim_tech);
-	ast_rtp_proto_unregister(&unistim_rtp);
+	ast_rtp_glue_unregister(&unistim_rtp_glue);
 
 	ast_mutex_lock(&monlock);
 	if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {

configs/sip.conf.sample

@@ -292,6 +292,8 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                        ; register their phones.
 
+;engine=asterisk                ; RTP engine to use when communicating with the device
+
 ;
 ; If regcontext is specified, Asterisk will dynamically create and destroy a
 ; NoOp priority 1 extension for a given peer who registers or unregisters with

include/asterisk/_private.h

@@ -41,6 +41,7 @@ int ast_tps_init(void); 		/*!< Provided by taskprocessor.c */
 int ast_timing_init(void);		/*!< Provided by timing.c */
 int ast_indications_init(void); /*!< Provided by indications.c */
 int ast_indications_reload(void);/*!< Provided by indications.c */
+void ast_stun_init(void);               /*!< Provided by stun.c */
 
 /*!
  * \brief Reload asterisk modules.

include/asterisk/rtp.h

@@ -1,416 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2006, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file rtp.h
- * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
- *
- * RTP is defined in RFC 3550.
- */
-
-#ifndef _ASTERISK_RTP_H
-#define _ASTERISK_RTP_H
-
-#include "asterisk/network.h"
-
-#include "asterisk/frame.h"
-#include "asterisk/io.h"
-#include "asterisk/sched.h"
-#include "asterisk/channel.h"
-#include "asterisk/linkedlists.h"
-
-#if defined(__cplusplus) || defined(c_plusplus)
-extern "C" {
-#endif
-
-/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
-/*! DTMF (RFC2833) */
-#define AST_RTP_DTMF            	(1 << 0)
-/*! 'Comfort Noise' (RFC3389) */
-#define AST_RTP_CN              	(1 << 1)
-/*! DTMF (Cisco Proprietary) */
-#define AST_RTP_CISCO_DTMF      	(1 << 2)
-/*! Maximum RTP-specific code */
-#define AST_RTP_MAX             	AST_RTP_CISCO_DTMF
-
-/*! Maxmum number of payload defintions for a RTP session */
-#define MAX_RTP_PT			256
-
-/*! T.140 Redundancy Maxium number of generations */
-#define RED_MAX_GENERATION 5
-
-#define FLAG_3389_WARNING		(1 << 0)
-
-enum ast_rtp_options {
-	AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
-};
-
-enum ast_rtp_get_result {
-	/*! Failed to find the RTP structure */
-	AST_RTP_GET_FAILED = 0,
-	/*! RTP structure exists but true native bridge can not occur so try partial */
-	AST_RTP_TRY_PARTIAL,
-	/*! RTP structure exists and native bridge can occur */
-	AST_RTP_TRY_NATIVE,
-};
-
-/*! \brief Variables used in ast_rtcp_get function */
-enum ast_rtp_qos_vars {
-	AST_RTP_TXCOUNT,
-	AST_RTP_RXCOUNT,
-	AST_RTP_TXJITTER,
-	AST_RTP_RXJITTER,
-	AST_RTP_RXPLOSS,
-	AST_RTP_TXPLOSS,
-	AST_RTP_RTT
-};
-
-struct ast_rtp;
-/*! T.140 Redundancy structure*/
-struct rtp_red;
-
-/*! \brief The value of each payload format mapping: */
-struct rtpPayloadType {
-	int isAstFormat;		/*!< whether the following code is an AST_FORMAT */
-	int code;
-};
-
-/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem 
-*/
-struct ast_rtp_protocol {
-	/*! Get RTP struct, or NULL if unwilling to transfer */
-	enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
-	/*! Get RTP struct, or NULL if unwilling to transfer */
-	enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
-	/*! Get RTP struct, or NULL if unwilling to transfer */
-	enum ast_rtp_get_result (* const get_trtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
-	/*! Set RTP peer */
-	int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, struct ast_rtp *tpeer, int codecs, int nat_active);
-	int (* const get_codec)(struct ast_channel *chan);
-	const char * const type;
-	AST_LIST_ENTRY(ast_rtp_protocol) list;
-};
-
-enum ast_rtp_quality_type {
-	RTPQOS_SUMMARY = 0,
-	RTPQOS_JITTER,
-	RTPQOS_LOSS,
-	RTPQOS_RTT
-};
-
-/*! \brief RTCP quality report storage */
-struct ast_rtp_quality {
-	unsigned int local_ssrc;          /*!< Our SSRC */
-	unsigned int local_lostpackets;   /*!< Our lost packets */
-	double       local_jitter;        /*!< Our calculated jitter */
-	unsigned int local_count;         /*!< Number of received packets */
-	unsigned int remote_ssrc;         /*!< Their SSRC */
-	unsigned int remote_lostpackets;  /*!< Their lost packets */
-	double       remote_jitter;       /*!< Their reported jitter */
-	unsigned int remote_count;        /*!< Number of transmitted packets */
-	double       rtt;                 /*!< Round trip time */
-};
-
-/*! RTP callback structure */
-typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
-
-/*!
- * \brief Get the amount of space required to hold an RTP session
- * \return number of bytes required
- */
-size_t ast_rtp_alloc_size(void);
-
-/*!
- * \brief Initializate a RTP session.
- *
- * \param sched
- * \param io
- * \param rtcpenable
- * \param callbackmode
- * \return A representation (structure) of an RTP session.
- */
-struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
-
-/*!
- * \brief Initializate a RTP session using an in_addr structure.
- *
- * This fuction gets called by ast_rtp_new().
- *
- * \param sched
- * \param io
- * \param rtcpenable
- * \param callbackmode
- * \param in
- * \return A representation (structure) of an RTP session.
- */
-struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
-
-void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
-
-/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
-int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
-
-void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
-
-struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
-
-/*! Destroy RTP session */
-void ast_rtp_destroy(struct ast_rtp *rtp);
-
-void ast_rtp_reset(struct ast_rtp *rtp);
-
-/*! Stop RTP session, do not destroy structure */
-void ast_rtp_stop(struct ast_rtp *rtp);
-
-void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
-
-void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
-
-int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
-
-struct ast_frame *ast_rtp_read(struct ast_rtp *rtp);
-
-struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp);
-
-int ast_rtp_fd(struct ast_rtp *rtp);
-
-int ast_rtcp_fd(struct ast_rtp *rtp);
-
-int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
-
-int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
-
-int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
-
-int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
-
-void ast_rtp_new_source(struct ast_rtp *rtp);
-
-/*! \brief  Setting RTP payload types from lines in a SDP description: */
-void ast_rtp_pt_clear(struct ast_rtp* rtp);
-/*! \brief Set payload types to defaults */
-void ast_rtp_pt_default(struct ast_rtp* rtp);
-
-/*! \brief Copy payload types between RTP structures */
-void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
-
-/*! \brief Activate payload type */
-void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
-
-/*! \brief clear payload type */
-void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
-
-/*! \brief Set payload type to a known MIME media type for a codec
- *
- * \param rtp RTP structure to modify
- * \param pt Payload type entry to modify
- * \param mimeType top-level MIME type of media stream (typically "audio", "video", "text", etc.)
- * \param mimeSubtype MIME subtype of media stream (typically a codec name)
- * \param options Zero or more flags from the ast_rtp_options enum
- *
- * This function 'fills in' an entry in the list of possible formats for
- * a media stream associated with an RTP structure.
- *
- * \retval 0 on success
- * \retval -1 if the payload type is out of range
- * \retval -2 if the mimeType/mimeSubtype combination was not found
- */
-int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
-			     char *mimeType, char *mimeSubtype,
-			     enum ast_rtp_options options);
-
-/*! \brief Set payload type to a known MIME media type for a codec with a specific sample rate
- *
- * \param rtp RTP structure to modify
- * \param pt Payload type entry to modify
- * \param mimeType top-level MIME type of media stream (typically "audio", "video", "text", etc.)
- * \param mimeSubtype MIME subtype of media stream (typically a codec name)
- * \param options Zero or more flags from the ast_rtp_options enum
- * \param sample_rate The sample rate of the media stream
- *
- * This function 'fills in' an entry in the list of possible formats for
- * a media stream associated with an RTP structure.
- *
- * \retval 0 on success
- * \retval -1 if the payload type is out of range
- * \retval -2 if the mimeType/mimeSubtype combination was not found
- */
-int ast_rtp_set_rtpmap_type_rate(struct ast_rtp* rtp, int pt,
-				 char *mimeType, char *mimeSubtype,
-				 enum ast_rtp_options options,
-				 unsigned int sample_rate);
-
-/*! \brief  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_get_current_formats(struct ast_rtp* rtp,
-				 int* astFormats, int* nonAstFormats);
-
-/*! \brief  Mapping an Asterisk code into a MIME subtype (string): */
-const char *ast_rtp_lookup_mime_subtype(int isAstFormat, int code,
-					enum ast_rtp_options options);
-
-/*! \brief Get the sample rate associated with known RTP payload types
- *
- * \param isAstFormat True if the value in the 'code' parameter is an AST_FORMAT value
- * \param code Format code, either from AST_FORMAT list or from AST_RTP list
- *
- * \return the sample rate if the format was found, zero if it was not found
- */
-unsigned int ast_rtp_lookup_sample_rate(int isAstFormat, int code);
-
-/*! \brief Build a string of MIME subtype names from a capability list */
-char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
-				   const int isAstFormat, enum ast_rtp_options options);
-
-void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
-
-int ast_rtp_getnat(struct ast_rtp *rtp);
-
-/*! \brief Indicate whether this RTP session is carrying DTMF or not */
-void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
-
-/*! \brief Compensate for devices that send RFC2833 packets all at once */
-void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
-
-/*! \brief Enable STUN capability */
-void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
-
-/*! \brief Generic STUN request
- * send a generic stun request to the server specified.
- * \param s the socket used to send the request
- * \param dst the address of the STUN server
- * \param username if non null, add the username in the request
- * \param answer if non null, the function waits for a response and
- *    puts here the externally visible address.
- * \return 0 on success, other values on error.
- * The interface it may change in the future.
- */
-int ast_stun_request(int s, struct sockaddr_in *dst,
-	const char *username, struct sockaddr_in *answer);
-
-/*! \brief Send STUN request for an RTP socket
- * Deprecated, this is just a wrapper for ast_rtp_stun_request()
- */
-void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
-
-/*! \brief The RTP bridge.
-	\arg \ref AstRTPbridge
-*/
-int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
-
-/*! \brief Register an RTP channel client */
-int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
-
-/*! \brief Unregister an RTP channel client */
-void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
-
-int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
-
-/*! \brief If possible, create an early bridge directly between the devices without
-           having to send a re-invite later */
-int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
-
-/*! \brief Get QOS stats on a RTP channel
- * \since 1.6.1
- */
-int ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen);
-
-/*! \brief Return RTP and RTCP QoS values
- * \since 1.6.1
- */
-unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value);
-
-/*! \brief Set RTPAUDIOQOS(...) variables on a channel when it is being hung up
- * \since 1.6.1
- */
-void ast_rtp_set_vars(struct ast_channel *chan, struct ast_rtp *rtp);
-
-/*! \brief Return RTCP quality string 
- *
- *  \param rtp An rtp structure to get qos information about.
- *
- *  \param qual An (optional) rtp quality structure that will be 
- *              filled with the quality information described in 
- *              the ast_rtp_quality structure. This structure is
- *              not dependent on any qtype, so a call for any
- *              type of information would yield the same results
- *              because ast_rtp_quality is not a data type 
- *              specific to any qos type.
- *
- *  \param qtype The quality type you'd like, default should be
- *               RTPQOS_SUMMARY which returns basic information
- *               about the call. The return from RTPQOS_SUMMARY
- *               is basically ast_rtp_quality in a string. The
- *               other types are RTPQOS_JITTER, RTPQOS_LOSS and
- *               RTPQOS_RTT which will return more specific 
- *               statistics.
- * \version 1.6.1 added qtype parameter
- */
-char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual, enum ast_rtp_quality_type qtype);
-/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message  in SIP */
-int ast_rtcp_send_h261fur(void *data);
-
-void ast_rtp_init(void);                                      /*! Initialize RTP subsystem */
-int ast_rtp_reload(void);                                     /*! reload rtp configuration */
-void ast_rtp_new_init(struct ast_rtp *rtp);
-
-/*! \brief Set codec preference */
-void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
-
-/*! \brief Get codec preference */
-struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
-
-/*! \brief get format from predefined dynamic payload format */
-int ast_rtp_codec_getformat(int pt);
-
-/*! \brief Set rtp timeout */
-void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
-/*! \brief Set rtp hold timeout */
-void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
-/*! \brief set RTP keepalive interval */
-void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
-/*! \brief Get RTP keepalive interval */
-int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
-/*! \brief Get rtp hold timeout */
-int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
-/*! \brief Get rtp timeout */
-int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
-/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
-void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
-
-/*! \brief Initalize t.140 redudancy 
- * \param ti time between each t140red frame is sent
- * \param red_pt payloadtype for RTP packet
- * \param pt payloadtype numbers for each generation including primary data
- * \param num_gen number of redundant generations, primary data excluded
- * \since 1.6.1
- */
-int ast_rtp_red_init(struct ast_rtp *rtp, int ti, int *pt, int num_gen);
-
-/*! \brief Buffer t.140 data */
-void ast_red_buffer_t140(struct ast_rtp *rtp, struct ast_frame *f);
-
-
-
-#if defined(__cplusplus) || defined(c_plusplus)
-}
-#endif
-
-#endif /* _ASTERISK_RTP_H */

include/asterisk/stun.h

@@ -0,0 +1,71 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file stun.h
+ * \brief STUN support.
+ *
+ * STUN is defined in RFC 3489.
+ */
+
+#ifndef _ASTERISK_STUN_H
+#define _ASTERISK_STUN_H
+
+#include "asterisk/network.h"
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+enum ast_stun_result {
+	AST_STUN_IGNORE = 0,
+	AST_STUN_ACCEPT,
+};
+
+struct stun_attr;
+
+/*! \brief Generic STUN request
+ * send a generic stun request to the server specified.
+ * \param s the socket used to send the request
+ * \param dst the address of the STUN server
+ * \param username if non null, add the username in the request
+ * \param answer if non null, the function waits for a response and
+ *    puts here the externally visible address.
+ * \return 0 on success, other values on error.
+ * The interface it may change in the future.
+ */
+int ast_stun_request(int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer);
+
+/*! \brief callback type to be invoked on stun responses. */
+typedef int (stun_cb_f)(struct stun_attr *attr, void *arg);
+
+/*! \brief handle an incoming STUN message.
+ *
+ * Do some basic sanity checks on packet size and content,
+ * try to extract a bit of information, and possibly reply.
+ * At the moment this only processes BIND requests, and returns
+ * the externally visible address of the request.
+ * If a callback is specified, invoke it with the attribute.
+ */
+int ast_stun_handle_packet(int s, struct sockaddr_in *src, unsigned char *data, size_t len, stun_cb_f *stun_cb, void *arg);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif /* _ASTERISK_STUN_H */

main/Makefile

@@ -20,7 +20,7 @@ include $(ASTTOPDIR)/Makefile.moddir_rules
 OBJS= tcptls.o io.o sched.o logger.o frame.o loader.o config.o channel.o \
 	translate.o file.o pbx.o cli.o md5.o term.o heap.o \
 	ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
-	cdr.o tdd.o acl.o rtp.o udptl.o manager.o asterisk.o \
+	cdr.o tdd.o acl.o udptl.o manager.o asterisk.o \
 	dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
 	astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o \
 	utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
@@ -29,7 +29,7 @@ OBJS= tcptls.o io.o sched.o logger.o frame.o loader.o config.o channel.o \
 	strcompat.o threadstorage.o dial.o event.o adsistub.o audiohook.o \
 	astobj2.o hashtab.o global_datastores.o version.o \
 	features.o taskprocessor.o timing.o datastore.o xml.o xmldoc.o \
-	strings.o bridging.o poll.o
+	strings.o bridging.o poll.o rtp_engine.o stun.o
 
 # we need to link in the objects statically, not as a library, because
 # otherwise modules will not have them available if none of the static

main/asterisk.c

@@ -120,7 +120,6 @@ int daemon(int, int);  /* defined in libresolv of all places */
 #include "asterisk/cdr.h"
 #include "asterisk/pbx.h"
 #include "asterisk/enum.h"
-#include "asterisk/rtp.h"
 #include "asterisk/http.h"
 #include "asterisk/udptl.h"
 #include "asterisk/app.h"
@@ -3579,7 +3578,6 @@ int main(int argc, char *argv[])
 		exit(1);
 	}
 
-	ast_rtp_init();
 	ast_dsp_init();
 	ast_udptl_init();
 

main/loader.c

@@ -43,7 +43,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/manager.h"
 #include "asterisk/cdr.h"
 #include "asterisk/enum.h"
-#include "asterisk/rtp.h"
 #include "asterisk/http.h"
 #include "asterisk/lock.h"
 #include "asterisk/features.h"
@@ -243,7 +242,6 @@ static struct reload_classes {
 	{ "extconfig",	read_config_maps },
 	{ "enum",	ast_enum_reload },
 	{ "manager",	reload_manager },
-	{ "rtp",	ast_rtp_reload },
 	{ "http",	ast_http_reload },
 	{ "logger",	logger_reload },
 	{ "features",	ast_features_reload },

main/stun.c

@@ -0,0 +1,475 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ *
+ * \brief STUN Support
+ *
+ * \author Mark Spencer <markster@digium.com>
+ *
+ * \note STUN is defined in RFC 3489.
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 124370 $")
+
+#include "asterisk/_private.h"
+#include "asterisk/stun.h"
+#include "asterisk/cli.h"
+#include "asterisk/utils.h"
+#include "asterisk/channel.h"
+
+static int stundebug;			/*!< Are we debugging stun? */
+
+/*!
+ * \brief STUN support code
+ *
+ * This code provides some support for doing STUN transactions.
+ * Eventually it should be moved elsewhere as other protocols
+ * than RTP can benefit from it - e.g. SIP.
+ * STUN is described in RFC3489 and it is based on the exchange
+ * of UDP packets between a client and one or more servers to
+ * determine the externally visible address (and port) of the client
+ * once it has gone through the NAT boxes that connect it to the
+ * outside.
+ * The simplest request packet is just the header defined in
+ * struct stun_header, and from the response we may just look at
+ * one attribute, STUN_MAPPED_ADDRESS, that we find in the response.
+ * By doing more transactions with different server addresses we
+ * may determine more about the behaviour of the NAT boxes, of
+ * course - the details are in the RFC.
+ *
+ * All STUN packets start with a simple header made of a type,
+ * length (excluding the header) and a 16-byte random transaction id.
+ * Following the header we may have zero or more attributes, each
+ * structured as a type, length and a value (whose format depends
+ * on the type, but often contains addresses).
+ * Of course all fields are in network format.
+ */
+
+typedef struct { unsigned int id[4]; } __attribute__((packed)) stun_trans_id;
+
+struct stun_header {
+	unsigned short msgtype;
+	unsigned short msglen;
+	stun_trans_id  id;
+	unsigned char ies[0];
+} __attribute__((packed));
+
+struct stun_attr {
+	unsigned short attr;
+	unsigned short len;
+	unsigned char value[0];
+} __attribute__((packed));
+
+/*
+ * The format normally used for addresses carried by STUN messages.
+ */
+struct stun_addr {
+	unsigned char unused;
+	unsigned char family;
+	unsigned short port;
+	unsigned int addr;
+} __attribute__((packed));
+
+/*! \brief STUN message types
+ * 'BIND' refers to transactions used to determine the externally
+ * visible addresses. 'SEC' refers to transactions used to establish
+ * a session key for subsequent requests.
+ * 'SEC' functionality is not supported here.
+ */
+
+#define STUN_BINDREQ	0x0001
+#define STUN_BINDRESP	0x0101
+#define STUN_BINDERR	0x0111
+#define STUN_SECREQ	0x0002
+#define STUN_SECRESP	0x0102
+#define STUN_SECERR	0x0112
+
+/*! \brief Basic attribute types in stun messages.
+ * Messages can also contain custom attributes (codes above 0x7fff)
+ */
+#define STUN_MAPPED_ADDRESS	0x0001
+#define STUN_RESPONSE_ADDRESS	0x0002
+#define STUN_CHANGE_REQUEST	0x0003
+#define STUN_SOURCE_ADDRESS	0x0004
+#define STUN_CHANGED_ADDRESS	0x0005
+#define STUN_USERNAME		0x0006
+#define STUN_PASSWORD		0x0007
+#define STUN_MESSAGE_INTEGRITY	0x0008
+#define STUN_ERROR_CODE		0x0009
+#define STUN_UNKNOWN_ATTRIBUTES	0x000a
+#define STUN_REFLECTED_FROM	0x000b
+
+/*! \brief helper function to print message names */
+static const char *stun_msg2str(int msg)
+{
+	switch (msg) {
+	case STUN_BINDREQ:
+		return "Binding Request";
+	case STUN_BINDRESP:
+		return "Binding Response";
+	case STUN_BINDERR:
+		return "Binding Error Response";
+	case STUN_SECREQ:
+		return "Shared Secret Request";
+	case STUN_SECRESP:
+		return "Shared Secret Response";
+	case STUN_SECERR:
+		return "Shared Secret Error Response";
+	}
+	return "Non-RFC3489 Message";
+}
+
+/*! \brief helper function to print attribute names */
+static const char *stun_attr2str(int msg)
+{
+	switch (msg) {
+	case STUN_MAPPED_ADDRESS:
+		return "Mapped Address";
+	case STUN_RESPONSE_ADDRESS:
+		return "Response Address";
+	case STUN_CHANGE_REQUEST:
+		return "Change Request";
+	case STUN_SOURCE_ADDRESS:
+		return "Source Address";
+	case STUN_CHANGED_ADDRESS:
+		return "Changed Address";
+	case STUN_USERNAME:
+		return "Username";
+	case STUN_PASSWORD:
+		return "Password";
+	case STUN_MESSAGE_INTEGRITY:
+		return "Message Integrity";
+	case STUN_ERROR_CODE:
+		return "Error Code";
+	case STUN_UNKNOWN_ATTRIBUTES:
+		return "Unknown Attributes";
+	case STUN_REFLECTED_FROM:
+		return "Reflected From";
+	}
+	return "Non-RFC3489 Attribute";
+}
+
+/*! \brief here we store credentials extracted from a message */
+struct stun_state {
+	const char *username;
+	const char *password;
+};
+
+static int stun_process_attr(struct stun_state *state, struct stun_attr *attr)
+{
+	if (stundebug)
+		ast_verbose("Found STUN Attribute %s (%04x), length %d\n",
+			    stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
+	switch (ntohs(attr->attr)) {
+	case STUN_USERNAME:
+		state->username = (const char *) (attr->value);
+		break;
+	case STUN_PASSWORD:
+		state->password = (const char *) (attr->value);
+		break;
+	default:
+		if (stundebug)
+			ast_verbose("Ignoring STUN attribute %s (%04x), length %d\n",
+				    stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
+	}
+	return 0;
+}
+
+/*! \brief append a string to an STUN message */
+static void append_attr_string(struct stun_attr **attr, int attrval, const char *s, int *len, int *left)
+{
+	int size = sizeof(**attr) + strlen(s);
+	if (*left > size) {
+		(*attr)->attr = htons(attrval);
+		(*attr)->len = htons(strlen(s));
+		memcpy((*attr)->value, s, strlen(s));
+		(*attr) = (struct stun_attr *)((*attr)->value + strlen(s));
+		*len += size;
+		*left -= size;
+	}
+}
+
+/*! \brief append an address to an STUN message */
+static void append_attr_address(struct stun_attr **attr, int attrval, struct sockaddr_in *sin, int *len, int *left)
+{
+	int size = sizeof(**attr) + 8;
+	struct stun_addr *addr;
+	if (*left > size) {
+		(*attr)->attr = htons(attrval);
+		(*attr)->len = htons(8);
+		addr = (struct stun_addr *)((*attr)->value);
+		addr->unused = 0;
+		addr->family = 0x01;
+		addr->port = sin->sin_port;
+		addr->addr = sin->sin_addr.s_addr;
+		(*attr) = (struct stun_attr *)((*attr)->value + 8);
+		*len += size;
+		*left -= size;
+	}
+}
+
+/*! \brief wrapper to send an STUN message */
+static int stun_send(int s, struct sockaddr_in *dst, struct stun_header *resp)
+{
+	return sendto(s, resp, ntohs(resp->msglen) + sizeof(*resp), 0,
+		      (struct sockaddr *)dst, sizeof(*dst));
+}
+
+/*! \brief helper function to generate a random request id */
+static void stun_req_id(struct stun_header *req)
+{
+	int x;
+	for (x = 0; x < 4; x++)
+		req->id.id[x] = ast_random();
+}
+
+/*! \brief handle an incoming STUN message.
+ *
+ * Do some basic sanity checks on packet size and content,
+ * try to extract a bit of information, and possibly reply.
+ * At the moment this only processes BIND requests, and returns
+ * the externally visible address of the request.
+ * If a callback is specified, invoke it with the attribute.
+ */
+int ast_stun_handle_packet(int s, struct sockaddr_in *src, unsigned char *data, size_t len, stun_cb_f *stun_cb, void *arg)
+{
+	struct stun_header *hdr = (struct stun_header *)data;
+	struct stun_attr *attr;
+	struct stun_state st;
+	int ret = AST_STUN_IGNORE;
+	int x;
+
+	/* On entry, 'len' is the length of the udp payload. After the
+	 * initial checks it becomes the size of unprocessed options,
+	 * while 'data' is advanced accordingly.
+	 */
+	if (len < sizeof(struct stun_header)) {
+		ast_debug(1, "Runt STUN packet (only %d, wanting at least %d)\n", (int) len, (int) sizeof(struct stun_header));
+		return -1;
+	}
+	len -= sizeof(struct stun_header);
+	data += sizeof(struct stun_header);
+	x = ntohs(hdr->msglen);	/* len as advertised in the message */
+	if (stundebug)
+		ast_verbose("STUN Packet, msg %s (%04x), length: %d\n", stun_msg2str(ntohs(hdr->msgtype)), ntohs(hdr->msgtype), x);
+	if (x > len) {
+		ast_debug(1, "Scrambled STUN packet length (got %d, expecting %d)\n", x, (int)len);
+	} else
+		len = x;
+	memset(&st, 0, sizeof(st));
+	while (len) {
+		if (len < sizeof(struct stun_attr)) {
+			ast_debug(1, "Runt Attribute (got %d, expecting %d)\n", (int)len, (int) sizeof(struct stun_attr));
+			break;
+		}
+		attr = (struct stun_attr *)data;
+		/* compute total attribute length */
+		x = ntohs(attr->len) + sizeof(struct stun_attr);
+		if (x > len) {
+			ast_debug(1, "Inconsistent Attribute (length %d exceeds remaining msg len %d)\n", x, (int)len);
+			break;
+		}
+		if (stun_cb)
+			stun_cb(attr, arg);
+		if (stun_process_attr(&st, attr)) {
+			ast_debug(1, "Failed to handle attribute %s (%04x)\n", stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr));
+			break;
+		}
+		/* Clear attribute id: in case previous entry was a string,
+		 * this will act as the terminator for the string.
+		 */
+		attr->attr = 0;
+		data += x;
+		len -= x;
+	}
+	/* Null terminate any string.
+	 * XXX NOTE, we write past the size of the buffer passed by the
+	 * caller, so this is potentially dangerous. The only thing that
+	 * saves us is that usually we read the incoming message in a
+	 * much larger buffer in the struct ast_rtp
+	 */
+	*data = '\0';
+
+	/* Now prepare to generate a reply, which at the moment is done
+	 * only for properly formed (len == 0) STUN_BINDREQ messages.
+	 */
+	if (len == 0) {
+		unsigned char respdata[1024];
+		struct stun_header *resp = (struct stun_header *)respdata;
+		int resplen = 0;	/* len excluding header */
+		int respleft = sizeof(respdata) - sizeof(struct stun_header);
+
+		resp->id = hdr->id;
+		resp->msgtype = 0;
+		resp->msglen = 0;
+		attr = (struct stun_attr *)resp->ies;
+		switch (ntohs(hdr->msgtype)) {
+		case STUN_BINDREQ:
+			if (stundebug)
+				ast_verbose("STUN Bind Request, username: %s\n",
+					    st.username ? st.username : "<none>");
+			if (st.username)
+				append_attr_string(&attr, STUN_USERNAME, st.username, &resplen, &respleft);
+			append_attr_address(&attr, STUN_MAPPED_ADDRESS, src, &resplen, &respleft);
+			resp->msglen = htons(resplen);
+			resp->msgtype = htons(STUN_BINDRESP);
+			stun_send(s, src, resp);
+			ret = AST_STUN_ACCEPT;
+			break;
+		default:
+			if (stundebug)
+				ast_verbose("Dunno what to do with STUN message %04x (%s)\n", ntohs(hdr->msgtype), stun_msg2str(ntohs(hdr->msgtype)));
+		}
+	}
+	return ret;
+}
+
+/*! \brief Extract the STUN_MAPPED_ADDRESS from the stun response.
+ * This is used as a callback for stun_handle_response
+ * when called from ast_stun_request.
+ */
+static int stun_get_mapped(struct stun_attr *attr, void *arg)
+{
+	struct stun_addr *addr = (struct stun_addr *)(attr + 1);
+	struct sockaddr_in *sa = (struct sockaddr_in *)arg;
+
+	if (ntohs(attr->attr) != STUN_MAPPED_ADDRESS || ntohs(attr->len) != 8)
+		return 1;	/* not us. */
+	sa->sin_port = addr->port;
+	sa->sin_addr.s_addr = addr->addr;
+	return 0;
+}
+
+/*! \brief Generic STUN request
+ * Send a generic stun request to the server specified,
+ * possibly waiting for a reply and filling the 'reply' field with
+ * the externally visible address. Note that in this case the request
+ * will be blocking.
+ * (Note, the interface may change slightly in the future).
+ *
+ * \param s the socket used to send the request
+ * \param dst the address of the STUN server
+ * \param username if non null, add the username in the request
+ * \param answer if non null, the function waits for a response and
+ *    puts here the externally visible address.
+ * \return 0 on success, other values on error.
+ */
+int ast_stun_request(int s, struct sockaddr_in *dst,
+	const char *username, struct sockaddr_in *answer)
+{
+	struct stun_header *req;
+	unsigned char reqdata[1024];
+	int reqlen, reqleft;
+	struct stun_attr *attr;
+	int res = 0;
+	int retry;
+
+	req = (struct stun_header *)reqdata;
+	stun_req_id(req);
+	reqlen = 0;
+	reqleft = sizeof(reqdata) - sizeof(struct stun_header);
+	req->msgtype = 0;
+	req->msglen = 0;
+	attr = (struct stun_attr *)req->ies;
+	if (username)
+		append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
+	req->msglen = htons(reqlen);
+	req->msgtype = htons(STUN_BINDREQ);
+	for (retry = 0; retry < 3; retry++) {	/* XXX make retries configurable */
+		/* send request, possibly wait for reply */
+		unsigned char reply_buf[1024];
+		fd_set rfds;
+		struct timeval to = { 3, 0 };	/* timeout, make it configurable */
+		struct sockaddr_in src;
+		socklen_t srclen;
+
+		res = stun_send(s, dst, req);
+		if (res < 0) {
+			ast_log(LOG_WARNING, "ast_stun_request send #%d failed error %d, retry\n",
+				retry, res);
+			continue;
+		}
+		if (answer == NULL)
+			break;
+		FD_ZERO(&rfds);
+		FD_SET(s, &rfds);
+		res = ast_select(s + 1, &rfds, NULL, NULL, &to);
+		if (res <= 0)	/* timeout or error */
+			continue;
+		memset(&src, 0, sizeof(src));
+		srclen = sizeof(src);
+		/* XXX pass -1 in the size, because stun_handle_packet might
+		 * write past the end of the buffer.
+		 */
+		res = recvfrom(s, reply_buf, sizeof(reply_buf) - 1,
+			0, (struct sockaddr *)&src, &srclen);
+		if (res < 0) {
+			ast_log(LOG_WARNING, "ast_stun_request recvfrom #%d failed error %d, retry\n",
+				retry, res);
+			continue;
+		}
+		memset(answer, 0, sizeof(struct sockaddr_in));
+		ast_stun_handle_packet(s, &src, reply_buf, res,
+			stun_get_mapped, answer);
+		res = 0; /* signal regular exit */
+		break;
+	}
+	return res;
+}
+
+static char *handle_cli_stun_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+	switch (cmd) {
+	case CLI_INIT:
+		e->command = "stun set debug {on|off}";
+		e->usage =
+			"Usage: stun set debug {on|off}\n"
+			"       Enable/Disable STUN (Simple Traversal of UDP through NATs)\n"
+			"       debugging\n";
+		return NULL;
+	case CLI_GENERATE:
+		return NULL;
+	}
+
+	if (a->argc != e->args)
+		return CLI_SHOWUSAGE;
+
+	if (!strncasecmp(a->argv[e->args-1], "on", 2))
+		stundebug = 1;
+	else if (!strncasecmp(a->argv[e->args-1], "off", 3))
+		stundebug = 0;
+	else
+		return CLI_SHOWUSAGE;
+
+	ast_cli(a->fd, "STUN Debugging %s\n", stundebug ? "Enabled" : "Disabled");
+	return CLI_SUCCESS;
+}
+
+static struct ast_cli_entry cli_stun[] = {
+	AST_CLI_DEFINE(handle_cli_stun_set_debug, "Enable/Disable STUN debugging"),
+};
+
+/*! \brief Initialize the STUN system in Asterisk */
+void ast_stun_init(void)
+{
+	ast_cli_register_multiple(cli_stun, sizeof(cli_stun) / sizeof(struct ast_cli_entry));
+}