Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 240 insertions, 94 deletions

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 738b58fae..b2543893a 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -39,6 +39,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/pbx.h"
 #include "asterisk/translate.h"
 #include "asterisk/netsock2.h"
+#include "asterisk/_private.h"
 
 struct ast_srtp_res *res_srtp = NULL;
 struct ast_srtp_policy_res *res_srtp_policy = NULL;
@@ -83,50 +84,14 @@ static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
 
 /*! The following array defines the MIME Media type (and subtype) for each
    of our codecs, or RTP-specific data type. */
-static const struct ast_rtp_mime_type {
+static struct ast_rtp_mime_type {
 	struct ast_rtp_payload_type payload_type;
 	char *type;
 	char *subtype;
 	unsigned int sample_rate;
-} ast_rtp_mime_types[] = {
-	{{1, {.id = AST_FORMAT_G723_1}, 0}, "audio", "G723", 8000},
-	{{1, {.id = AST_FORMAT_GSM}, 0}, "audio", "GSM", 8000},
-	{{1, {.id = AST_FORMAT_ULAW}, 0}, "audio", "PCMU", 8000},
-	{{1, {.id = AST_FORMAT_ULAW}, 0}, "audio", "G711U", 8000},
-	{{1, {.id = AST_FORMAT_ALAW}, 0}, "audio", "PCMA", 8000},
-	{{1, {.id = AST_FORMAT_ALAW}, 0}, "audio", "G711A", 8000},
-	{{1, {.id = AST_FORMAT_G726}, 0}, "audio", "G726-32", 8000},
-	{{1, {.id = AST_FORMAT_ADPCM}, 0}, "audio", "DVI4", 8000},
-	{{1, {.id = AST_FORMAT_SLINEAR}, 0}, "audio", "L16", 8000},
-	{{1, {.id = AST_FORMAT_SLINEAR16}, 0}, "audio", "L16", 16000},
-	{{1, {.id = AST_FORMAT_LPC10}, 0}, "audio", "LPC", 8000},
-	{{1, {.id = AST_FORMAT_G729A}, 0}, "audio", "G729", 8000},
-	{{1, {.id = AST_FORMAT_G729A}, 0}, "audio", "G729A", 8000},
-	{{1, {.id = AST_FORMAT_G729A}, 0}, "audio", "G.729", 8000},
-	{{1, {.id = AST_FORMAT_SPEEX}, 0}, "audio", "speex", 8000},
-	{{1, {.id = AST_FORMAT_SPEEX16}, 0},  "audio", "speex", 16000},
-	{{1, {.id = AST_FORMAT_ILBC}, 0}, "audio", "iLBC", 8000},
-	/* this is the sample rate listed in the RTP profile for the G.722
-	              codec, *NOT* the actual sample rate of the media stream
-	*/
-	{{1, {.id = AST_FORMAT_G722}, 0}, "audio", "G722", 8000},
-	{{1, {.id = AST_FORMAT_G726_AAL2}, 0}, "audio", "AAL2-G726-32", 8000},
-	{{0, {.id = 0}, AST_RTP_DTMF}, "audio", "telephone-event", 8000},
-	{{0, {.id = 0}, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event", 8000},
-	{{0, {.id = 0}, AST_RTP_CN}, "audio", "CN", 8000},
-	{{1, {.id = AST_FORMAT_JPEG}, 0}, "video", "JPEG", 90000},
-	{{1, {.id = AST_FORMAT_PNG}, 0}, "video", "PNG", 90000},
-	{{1, {.id = AST_FORMAT_H261}, 0}, "video", "H261", 90000},
-	{{1, {.id = AST_FORMAT_H263}, 0}, "video", "H263", 90000},
-	{{1, {.id = AST_FORMAT_H263_PLUS}, 0}, "video", "h263-1998", 90000},
-	{{1, {.id = AST_FORMAT_H264}, 0}, "video", "H264", 90000},
-	{{1, {.id = AST_FORMAT_MP4_VIDEO}, 0}, "video", "MP4V-ES", 90000},
-	{{1, {.id = AST_FORMAT_T140RED}, 0}, "text", "RED", 1000},
-	{{1, {.id = AST_FORMAT_T140}, 0}, "text", "T140", 1000},
-	{{1, {.id = AST_FORMAT_SIREN7}, 0}, "audio", "G7221", 16000},
-	{{1, {.id = AST_FORMAT_SIREN14}, 0}, "audio", "G7221", 32000},
-	{{1, {.id = AST_FORMAT_G719}, 0}, "audio", "G719", 48000},
-};
+} ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
+static ast_rwlock_t mime_types_lock;
+static int mime_types_len = 0;
 
 /*!
  * \brief Mapping between Asterisk codecs and rtp payload types
@@ -138,46 +103,8 @@ static const struct ast_rtp_mime_type {
  * See http://www.iana.org/assignments/rtp-parameters for a list of
  * assigned values
  */
-static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
-	[0] = {1, {.id = AST_FORMAT_ULAW}, 0},
-	#ifdef USE_DEPRECATED_G726
-	[2] = {1, {.id = AST_FORMAT_G726}, 0},/* Technically this is G.721, but if Cisco can do it, so can we... */
-	#endif
-	[3] = {1, {.id = AST_FORMAT_GSM}, 0},
-	[4] = {1, {.id = AST_FORMAT_G723_1}, 0},
-	[5] = {1, {.id = AST_FORMAT_ADPCM}, 0},/* 8 kHz */
-	[6] = {1, {.id = AST_FORMAT_ADPCM}, 0}, /* 16 kHz */
-	[7] = {1, {.id = AST_FORMAT_LPC10}, 0},
-	[8] = {1, {.id = AST_FORMAT_ALAW}, 0},
-	[9] = {1, {.id = AST_FORMAT_G722}, 0},
-	[10] = {1, {.id = AST_FORMAT_SLINEAR}, 0}, /* 2 channels */
-	[11] = {1, {.id = AST_FORMAT_SLINEAR}, 0}, /* 1 channel */
-	[13] = {0, {.id = 0}, AST_RTP_CN},
-	[16] = {1, {.id = AST_FORMAT_ADPCM}, 0}, /* 11.025 kHz */
-	[17] = {1, {.id = AST_FORMAT_ADPCM}, 0}, /* 22.050 kHz */
-	[18] = {1, {.id = AST_FORMAT_G729A}, 0},
-	[19] = {0, {.id = 0}, AST_RTP_CN},         /* Also used for CN */
-	[26] = {1, {.id = AST_FORMAT_JPEG}, 0},
-	[31] = {1, {.id = AST_FORMAT_H261}, 0},
-	[34] = {1, {.id = AST_FORMAT_H263}, 0},
-	[97] = {1, {.id = AST_FORMAT_ILBC}, 0},
-	[98] = {1, {.id = AST_FORMAT_H263_PLUS}, 0},
-	[99] = {1, {.id = AST_FORMAT_H264}, 0},
-	[101] = {0, {.id = 0}, AST_RTP_DTMF},
-	[102] = {1, {.id = AST_FORMAT_SIREN7}, 0},
-	[103] = {1, {.id = AST_FORMAT_H263_PLUS}, 0},
-	[104] = {1, {.id = AST_FORMAT_MP4_VIDEO}, 0},
-	[105] = {1, {.id = AST_FORMAT_T140RED}, 0},   /* Real time text chat (with redundancy encoding) */
-	[106] = {1, {.id = AST_FORMAT_T140}, 0},      /* Real time text chat */
-	[110] = {1, {.id = AST_FORMAT_SPEEX}, 0},
-	[111] = {1, {.id = AST_FORMAT_G726}, 0},
-	[112] = {1, {.id = AST_FORMAT_G726_AAL2}, 0},
-	[115] = {1, {.id = AST_FORMAT_SIREN14}, 0},
-	[116] = {1, {.id = AST_FORMAT_G719}, 0},
-	[117] = {1, {.id = AST_FORMAT_SPEEX16}, 0},
-	[118] = {1, {.id = AST_FORMAT_SLINEAR16}, 0}, /* 16 Khz signed linear */
-	[121] = {0, {.id = 0}, AST_RTP_CISCO_DTMF},   /* Must be type 121 */
-};
+static struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT];
+static ast_rwlock_t static_RTP_PT_lock;
 
 int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
 {
@@ -497,6 +424,7 @@ void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_r
 {
 	int i;
 
+	ast_rwlock_rdlock(&static_RTP_PT_lock);
 	for (i = 0; i < AST_RTP_MAX_PT; i++) {
 		if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
 
@@ -508,6 +436,7 @@ void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_r
 			}
 		}
 	}
+	ast_rwlock_unlock(&static_RTP_PT_lock);
 }
 
 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
@@ -529,7 +458,10 @@ void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_cod
 
 void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
 {
+
+	ast_rwlock_rdlock(&static_RTP_PT_lock);
 	if (payload < 0 || payload >= AST_RTP_MAX_PT || (!static_RTP_PT[payload].rtp_code && !static_RTP_PT[payload].asterisk_format)) {
+		ast_rwlock_unlock(&static_RTP_PT_lock);
 		return;
 	}
 
@@ -542,6 +474,7 @@ void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct as
 	if (instance && instance->engine && instance->engine->payload_set) {
 		instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, &codecs->payloads[payload].format, codecs->payloads[payload].rtp_code);
 	}
+	ast_rwlock_unlock(&static_RTP_PT_lock);
 }
 
 int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
@@ -555,7 +488,8 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 	if (pt < 0 || pt >= AST_RTP_MAX_PT)
 		return -1; /* bogus payload type */
 
-	for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
+	ast_rwlock_rdlock(&mime_types_lock);
+	for (i = 0; i < mime_types_len; ++i) {
 		const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
 
 		if (strcasecmp(mimesubtype, t->subtype)) {
@@ -587,6 +521,7 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 
 		break;
 	}
+	ast_rwlock_unlock(&mime_types_lock);
 
 	return (found ? 0 : -2);
 }
@@ -626,12 +561,26 @@ struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs
 	ast_format_copy(&result.format, &codecs->payloads[payload].format);
 
 	if (!result.rtp_code && !result.asterisk_format) {
+		ast_rwlock_rdlock(&static_RTP_PT_lock);
 		result = static_RTP_PT[payload];
+		ast_rwlock_unlock(&static_RTP_PT_lock);
 	}
 
 	return result;
 }
 
+
+struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
+{
+	if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+		return NULL;
+	}
+	if (!codecs->payloads[payload].asterisk_format) {
+		return NULL;
+	}
+	return &codecs->payloads[payload].format;
+}
+
 void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
 {
 	int i;
@@ -654,7 +603,7 @@ void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_fo
 int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
 {
 	int i;
-
+	int res = -1;
 	for (i = 0; i < AST_RTP_MAX_PT; i++) {
 		if (codecs->payloads[i].asterisk_format && asterisk_format && format &&
 			(ast_format_cmp(format, &codecs->payloads[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
@@ -665,56 +614,71 @@ int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_form
 		}
 	}
 
+	ast_rwlock_rdlock(&static_RTP_PT_lock);
 	for (i = 0; i < AST_RTP_MAX_PT; i++) {
 		if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
 			(ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
-			return i;
+			res = i;
+			break;
 		} else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
 			(static_RTP_PT[i].rtp_code == code)) {
-			return i;
+			res = i;
+			break;
 		}
 	}
+	ast_rwlock_unlock(&static_RTP_PT_lock);
 
-	return -1;
+	return res;
 }
 
 const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
 {
 	int i;
+	const char *res = "";
 
-	for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); i++) {
+	ast_rwlock_rdlock(&mime_types_lock);
+	for (i = 0; i < mime_types_len; i++) {
 		if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
 			(ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
 			if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
-				return "G726-32";
+				res = "G726-32";
+				break;
 			} else {
-				return ast_rtp_mime_types[i].subtype;
+				res = ast_rtp_mime_types[i].subtype;
+				break;
 			}
 		} else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
 			ast_rtp_mime_types[i].payload_type.rtp_code == code) {
 
-			return ast_rtp_mime_types[i].subtype;
+			res = ast_rtp_mime_types[i].subtype;
+			break;
 		}
 	}
+	ast_rwlock_unlock(&mime_types_lock);
 
-	return "";
+	return res;
 }
 
 unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
 {
 	unsigned int i;
+	unsigned int res = 0;
 
-	for (i = 0; i < ARRAY_LEN(ast_rtp_mime_types); ++i) {
+	ast_rwlock_rdlock(&mime_types_lock);
+	for (i = 0; i < mime_types_len; ++i) {
 		if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
 			(ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
-			return ast_rtp_mime_types[i].sample_rate;
+			res = ast_rtp_mime_types[i].sample_rate;
+			break;
 		} else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
 			ast_rtp_mime_types[i].payload_type.rtp_code == code) {
-			return ast_rtp_mime_types[i].sample_rate;
+			res = ast_rtp_mime_types[i].sample_rate;
+			break;
 		}
 	}
+	ast_rwlock_unlock(&mime_types_lock);
 
-	return 0;
+	return res;
 }
 
 char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, struct ast_format_cap *ast_format_capability, int rtp_capability, const int asterisk_format, enum ast_rtp_options options)
@@ -1879,3 +1843,185 @@ struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
 {
 	return instance->srtp;
 }
+
+static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
+{
+	int x = mime_types_len;
+	if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
+		return;
+	}
+
+	ast_rwlock_wrlock(&mime_types_lock);
+	if (format) {
+		ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
+		ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
+	} else {
+		ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
+	}
+	ast_rtp_mime_types[x].type = type;
+	ast_rtp_mime_types[x].subtype = subtype;
+	ast_rtp_mime_types[x].sample_rate = sample_rate;
+	mime_types_len++;
+	ast_rwlock_unlock(&mime_types_lock);
+}
+
+static void add_static_payload(int map, const struct ast_format *format, int rtp_code)
+{
+	int x;
+	ast_rwlock_wrlock(&static_RTP_PT_lock);
+	if (map < 0) {
+		/* find next available dynamic payload slot */
+		for (x = 96; x < 127; x++) {
+			if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
+				map = x;
+				break;
+			}
+		}
+	}
+
+	if (map < 0) {
+		ast_log(LOG_WARNING, "No Dynamic RTP mapping avaliable for format %s\n" ,ast_getformatname(format));
+		ast_rwlock_unlock(&static_RTP_PT_lock);
+		return;
+	}
+
+	if (format) {
+		static_RTP_PT[map].asterisk_format = 1;
+		ast_format_copy(&static_RTP_PT[map].format, format);
+	} else {
+		static_RTP_PT[map].rtp_code = rtp_code;
+	}
+	ast_rwlock_unlock(&static_RTP_PT_lock);
+}
+
+int ast_rtp_engine_load_format(const struct ast_format *format)
+{
+	switch (format->id) {
+	case AST_FORMAT_SILK:
+		set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
+		add_static_payload(-1, format, 0);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+int ast_rtp_engine_unload_format(const struct ast_format *format)
+{
+	int x;
+	int y = 0;
+
+	ast_rwlock_wrlock(&static_RTP_PT_lock);
+	/* remove everything pertaining to this format id from the lists */
+	for (x = 0; x < AST_RTP_MAX_PT; x++) {
+		if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
+			memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
+		}
+	}
+	ast_rwlock_unlock(&static_RTP_PT_lock);
+
+
+	ast_rwlock_wrlock(&mime_types_lock);
+	/* rebuild the list skipping the items matching this id */
+	for (x = 0; x < mime_types_len; x++) {
+		if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
+			continue;
+		}
+		ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
+		y++;
+	}
+	mime_types_len = y;
+	ast_rwlock_unlock(&mime_types_lock);
+	return 0;
+}
+
+int ast_rtp_engine_init()
+{
+	struct ast_format tmpfmt;
+
+	ast_rwlock_init(&mime_types_lock);
+	ast_rwlock_init(&static_RTP_PT_lock);
+
+	/* Define all the RTP mime types available */
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0,  "audio", "speex", 16000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0,  "audio", "speex", 32000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0, "audio", "iLBC", 8000);
+	/* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0, "audio", "AAL2-G726-32", 8000);
+	set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
+	set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
+	set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "h263-1998", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
+
+	/* Define the static rtp payload mappings */
+	add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
+	#ifdef USE_DEPRECATED_G726
+	add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
+	#endif
+	add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
+	add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
+	add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
+	add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
+	add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
+	add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
+	add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
+	add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
+	add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
+	add_static_payload(13, NULL, AST_RTP_CN);
+	add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
+	add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
+	add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
+	add_static_payload(19, NULL, AST_RTP_CN);         /* Also used for CN */
+	add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
+	add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
+	add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
+	add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
+	add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
+	add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
+	add_static_payload(101, NULL, AST_RTP_DTMF);
+	add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
+	add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
+	add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
+	add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0);   /* Real time text chat (with redundancy encoding) */
+	add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0);     /* Real time text chat */
+	add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
+	add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
+	add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
+	add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
+	add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
+	add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
+	add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
+	add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
+	add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
+
+	return 0;
+}