src/libosmo-mgcp/mgcp_sdp.c - osmo-mgw - Gitiles

 /*
  * Some SDP file parsing...
  *
  * (C) 2009-2015 by Holger Hans Peter Freyther <zecke@selfish.org>
  * (C) 2009-2014 by On-Waves
  * All Rights Reserved
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU Affero General Public License as published by
  * the Free Software Foundation; either version 3 of the License, or
  * (at your option) any later version.
  *
  * This program is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  * GNU Affero General Public License for more details.
  *
  * You should have received a copy of the GNU Affero General Public License
  * along with this program.  If not, see <http://www.gnu.org/licenses/>.
  *
  */

 #include <osmocom/core/msgb.h>
 #include <osmocom/mgcp/mgcp.h>
 #include <osmocom/mgcp/mgcp_internal.h>
 #include <osmocom/mgcp/mgcp_msg.h>
 #include <osmocom/mgcp/mgcp_endp.h>

 #include <errno.h>

 struct sdp_rtp_map {
 	/* the type */
 	int payload_type;
 	/* null, static or later dynamic codec name */
 	char *codec_name;
 	/* A pointer to the original line for later parsing */
 	char *map_line;

 	int rate;
 	int channels;
 };

 /*! Set codec configuration depending on payload type and codec name.
  *  \param[in] ctx talloc context.
  *  \param[out] codec configuration (caller provided memory).
  *  \param[in] payload_type codec type id (e.g. 3 for GSM, -1 when undefined).
  *  \param[in] audio_name audio codec name (e.g. "GSM/8000/1").
  *  \returns 0 on success, -1 on failure. */
 int mgcp_set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
 			int payload_type, const char *audio_name)
 {
 	int rate = codec->rate;
 	int channels = codec->channels;
 	char audio_codec[64];

 	talloc_free(codec->subtype_name);
 	codec->subtype_name = NULL;
 	talloc_free(codec->audio_name);
 	codec->audio_name = NULL;

 	if (payload_type != PTYPE_UNDEFINED)
 		codec->payload_type = payload_type;

 	if (!audio_name) {
 		switch (payload_type) {
 		case 0:
 			audio_name = "PCMU/8000/1";
 			break;
 		case 3:
 			audio_name = "GSM/8000/1";
 			break;
 		case 8:
 			audio_name = "PCMA/8000/1";
 			break;
 		case 18:
 			audio_name = "G729/8000/1";
 			break;
 		default:
 			/* Payload type is unknown, don't change rate and
 			 * channels. */
 			/* TODO: return value? */
 			return 0;
 		}
 	}

 	if (sscanf(audio_name, "%63[^/]/%d/%d",
 		   audio_codec, &rate, &channels) < 1)
 		return -EINVAL;

 	codec->rate = rate;
 	codec->channels = channels;
 	codec->subtype_name = talloc_strdup(ctx, audio_codec);
 	codec->audio_name = talloc_strdup(ctx, audio_name);

 	if (!strcmp(audio_codec, "G729")) {
 		codec->frame_duration_num = 10;
 		codec->frame_duration_den = 1000;
 	} else {
 		codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
 		codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
 	}

 	if (payload_type < 0) {
 		payload_type = 96;
 		if (rate == 8000 && channels == 1) {
 			if (!strcmp(audio_codec, "GSM"))
 				payload_type = 3;
 			else if (!strcmp(audio_codec, "PCMA"))
 				payload_type = 8;
 			else if (!strcmp(audio_codec, "PCMU"))
 				payload_type = 0;
 			else if (!strcmp(audio_codec, "G729"))
 				payload_type = 18;
 		}

 		codec->payload_type = payload_type;
 	}

 	if (channels != 1)
 		LOGP(DLMGCP, LOGL_NOTICE,
 		     "Channels != 1 in SDP: '%s'\n", audio_name);

 	return 0;
 }

 static void codecs_initialize(void *ctx, struct sdp_rtp_map *codecs, int used)
 {
 	int i;

 	for (i = 0; i < used; ++i) {
 		switch (codecs[i].payload_type) {
 		case 0:
 			codecs[i].codec_name = "PCMU";
 			codecs[i].rate = 8000;
 			codecs[i].channels = 1;
 			break;
 		case 3:
 			codecs[i].codec_name = "GSM";
 			codecs[i].rate = 8000;
 			codecs[i].channels = 1;
 			break;
 		case 8:
 			codecs[i].codec_name = "PCMA";
 			codecs[i].rate = 8000;
 			codecs[i].channels = 1;
 			break;
 		case 18:
 			codecs[i].codec_name = "G729";
 			codecs[i].rate = 8000;
 			codecs[i].channels = 1;
 			break;
 		}
 	}
 }

 static void codecs_update(void *ctx, struct sdp_rtp_map *codecs, int used,
 			  int payload, const char *audio_name)
 {
 	int i;

 	for (i = 0; i < used; ++i) {
 		char audio_codec[64];
 		int rate = -1;
 		int channels = -1;
 		if (codecs[i].payload_type != payload)
 			continue;
 		if (sscanf(audio_name, "%63[^/]/%d/%d",
 			   audio_codec, &rate, &channels) < 1) {
 			LOGP(DLMGCP, LOGL_ERROR, "Failed to parse '%s'\n",
 			     audio_name);
 			continue;
 		}

 		codecs[i].map_line = talloc_strdup(ctx, audio_name);
 		codecs[i].codec_name = talloc_strdup(ctx, audio_codec);
 		codecs[i].rate = rate;
 		codecs[i].channels = channels;
 		return;
 	}

 	LOGP(DLMGCP, LOGL_ERROR, "Unconfigured PT(%d) with %s\n", payload,
 	     audio_name);
 }

 /* Check if the codec matches what is set up in the trunk config */
 static int is_codec_compatible(const struct mgcp_endpoint *endp,
 			       const struct sdp_rtp_map *codec)
 {
 	char *codec_str;
 	char audio_codec[64];

 	if (!codec->codec_name)
 		return 0;

 	/* GSM, GSM/8000 and GSM/8000/1 should all be compatible...
 	 * let's go by name first. */
 	codec_str = endp->tcfg->audio_name;
 	if (sscanf(codec_str, "%63[^/]/%*d/%*d", audio_codec) < 1)
 		return 0;

 	return strcasecmp(audio_codec, codec->codec_name) == 0;
 }

 /*! Analyze SDP input string.
  *  \param[in] endp trunk endpoint.
  *  \param[out] conn associated rtp connection.
  *  \param[out] caller provided memory to store the parsing results.
  *  \returns 0 on success, -1 on failure.
  *
  *  Note: In conn (conn->end) the function returns the packet duration,
  *  the rtp port and the rtcp port */
 int mgcp_parse_sdp_data(const struct mgcp_endpoint *endp,
 			struct mgcp_conn_rtp *conn,
 			struct mgcp_parse_data *p)
 {
 	struct sdp_rtp_map codecs[10];
 	int codecs_used = 0;
 	char *line;
 	int maxptime = -1;
 	int i;
 	int codecs_assigned = 0;
 	void *tmp_ctx = talloc_new(NULL);
 	struct mgcp_rtp_end *rtp;

 	int payload;
 	int ptime, ptime2 = 0;
 	char audio_name[64];
 	int port, rc;
 	char ipv4[16];

 	OSMO_ASSERT(endp);
 	OSMO_ASSERT(conn);
 	OSMO_ASSERT(p);

 	rtp = &conn->end;
 	memset(&codecs, 0, sizeof(codecs));

 	for_each_line(line, p->save) {
 		switch (line[0]) {
 		case 'o':
 		case 's':
 		case 't':
 		case 'v':
 			/* skip these SDP attributes */
 			break;
 		case 'a':
 			if (sscanf(line, "a=rtpmap:%d %63s",
 				   &payload, audio_name) == 2) {
 				codecs_update(tmp_ctx, codecs,
 					      codecs_used, payload, audio_name);
 			} else
 			    if (sscanf
 				(line, "a=ptime:%d-%d", &ptime, &ptime2) >= 1) {
 				if (ptime2 > 0 && ptime2 != ptime)
 					rtp->packet_duration_ms = 0;
 				else
 					rtp->packet_duration_ms = ptime;
 			} else if (sscanf(line, "a=maxptime:%d", &ptime2)
 				   == 1) {
 				maxptime = ptime2;
 			}
 			break;
 		case 'm':
 			rc = sscanf(line,
 				    "m=audio %d RTP/AVP %d %d %d %d %d %d %d %d %d %d",
 				    &port, &codecs[0].payload_type,
 				    &codecs[1].payload_type,
 				    &codecs[2].payload_type,
 				    &codecs[3].payload_type,
 				    &codecs[4].payload_type,
 				    &codecs[5].payload_type,
 				    &codecs[6].payload_type,
 				    &codecs[7].payload_type,
 				    &codecs[8].payload_type,
 				    &codecs[9].payload_type);
 			if (rc >= 2) {
 				rtp->rtp_port = htons(port);
 				rtp->rtcp_port = htons(port + 1);
 				codecs_used = rc - 1;
 				codecs_initialize(tmp_ctx, codecs, codecs_used);
 			}
 			break;
 		case 'c':

 			if (sscanf(line, "c=IN IP4 %15s", ipv4) == 1) {
 				inet_aton(ipv4, &rtp->addr);
 			}
 			break;
 		default:
 			if (p->endp)
 				LOGP(DLMGCP, LOGL_NOTICE,
 				     "Unhandled SDP option: '%c'/%d on 0x%x\n",
 				     line[0], line[0],
 				     ENDPOINT_NUMBER(p->endp));
 			else
 				LOGP(DLMGCP, LOGL_NOTICE,
 				     "Unhandled SDP option: '%c'/%d\n",
 				     line[0], line[0]);
 			break;
 		}
 	}

 	/* Now select the primary and alt_codec */
 	for (i = 0; i < codecs_used && codecs_assigned < 2; ++i) {
 		struct mgcp_rtp_codec *codec = codecs_assigned == 0 ?
 		    &rtp->codec : &rtp->alt_codec;

 		if (endp->tcfg->no_audio_transcoding &&
 		    !is_codec_compatible(endp, &codecs[i])) {
 			LOGP(DLMGCP, LOGL_NOTICE, "Skipping codec %s\n",
 			     codecs[i].codec_name);
 			continue;
 		}

 		mgcp_set_audio_info(p->cfg, codec,
 				    codecs[i].payload_type, codecs[i].map_line);
 		codecs_assigned += 1;
 	}

 	if (codecs_assigned > 0) {
 		/* TODO/XXX: Store this per codec and derive it on use */
 		if (maxptime >= 0 && maxptime * rtp->codec.frame_duration_den >
 		    rtp->codec.frame_duration_num * 1500) {
 			/* more than 1 frame */
 			rtp->packet_duration_ms = 0;
 		}

 		LOGP(DLMGCP, LOGL_NOTICE,
 		     "Got media info via SDP: port %d, payload %d (%s), "
 		     "duration %d, addr %s\n",
 		     ntohs(rtp->rtp_port), rtp->codec.payload_type,
 		     rtp->codec.subtype_name ? rtp->
 		     codec.subtype_name : "unknown", rtp->packet_duration_ms,
 		     inet_ntoa(rtp->addr));
 	}

 	talloc_free(tmp_ctx);
 	return codecs_assigned > 0;
 }

 /*! Generate SDP response string.
  *  \param[in] endp trunk endpoint.
  *  \param[in] conn associated rtp connection.
  *  \param[out] sdp msg buffer to append resulting SDP string data.
  *  \param[in] addr IPV4 address string (e.g. 192.168.100.1).
  *  \returns 0 on success, -1 on failure. */
 int mgcp_write_response_sdp(const struct mgcp_endpoint *endp,
 			    const struct mgcp_conn_rtp *conn, struct msgb *sdp,
 			    const char *addr)
 {
 	const char *fmtp_extra;
 	const char *audio_name;
 	int payload_type;
 	int rc;

 	OSMO_ASSERT(endp);
 	OSMO_ASSERT(conn);
 	OSMO_ASSERT(sdp);
 	OSMO_ASSERT(addr);

 	/* FIXME: constify endp and conn args in get_net_donwlink_format_cb() */
 	endp->cfg->get_net_downlink_format_cb((struct mgcp_endpoint *)endp,
 					      &payload_type, &audio_name,
 					      &fmtp_extra,
 					      (struct mgcp_conn_rtp *)conn);

 	rc = msgb_printf(sdp,
 			 "v=0\r\n"
 			 "o=- %s 23 IN IP4 %s\r\n"
 			 "s=-\r\n"
 			 "c=IN IP4 %s\r\n"
 			 "t=0 0\r\n", conn->conn->id, addr, addr);

 	if (rc < 0)
 		goto buffer_too_small;

 	if (payload_type >= 0) {
 		rc = msgb_printf(sdp, "m=audio %d RTP/AVP %d\r\n",
 				 conn->end.local_port, payload_type);
 		if (rc < 0)
 			goto buffer_too_small;

 		if (audio_name && endp->tcfg->audio_send_name) {
 			rc = msgb_printf(sdp, "a=rtpmap:%d %s\r\n",
 					 payload_type, audio_name);

 			if (rc < 0)
 				goto buffer_too_small;
 		}

 		if (fmtp_extra) {
 			rc = msgb_printf(sdp, "%s\r\n", fmtp_extra);

 			if (rc < 0)
 				goto buffer_too_small;
 		}
 	}
 	if (conn->end.packet_duration_ms > 0 && endp->tcfg->audio_send_ptime) {
 		rc = msgb_printf(sdp, "a=ptime:%u\r\n",
 				 conn->end.packet_duration_ms);
 		if (rc < 0)
 			goto buffer_too_small;
 	}

 	return 0;

 buffer_too_small:
 	LOGP(DLMGCP, LOGL_ERROR, "SDP messagebuffer too small\n");
 	return -1;
 }
	/*
	* Some SDP file parsing...
	*
	* (C) 2009-2015 by Holger Hans Peter Freyther <zecke@selfish.org>
	* (C) 2009-2014 by On-Waves
	* All Rights Reserved
	*
	* This program is free software; you can redistribute it and/or modify
	* it under the terms of the GNU Affero General Public License as published by
	* the Free Software Foundation; either version 3 of the License, or
	* (at your option) any later version.
	*
	* This program is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	* GNU Affero General Public License for more details.
	*
	* You should have received a copy of the GNU Affero General Public License
	* along with this program. If not, see <http://www.gnu.org/licenses/>.
	*
	*/

	#include <osmocom/core/msgb.h>
	#include <osmocom/mgcp/mgcp.h>
	#include <osmocom/mgcp/mgcp_internal.h>
	#include <osmocom/mgcp/mgcp_msg.h>
	#include <osmocom/mgcp/mgcp_endp.h>

	#include <errno.h>

	struct sdp_rtp_map {
	/* the type */
	int payload_type;
	/* null, static or later dynamic codec name */
	char *codec_name;
	/* A pointer to the original line for later parsing */
	char *map_line;

	int rate;
	int channels;
	};

	/*! Set codec configuration depending on payload type and codec name.
	* \param[in] ctx talloc context.
	* \param[out] codec configuration (caller provided memory).
	* \param[in] payload_type codec type id (e.g. 3 for GSM, -1 when undefined).
	* \param[in] audio_name audio codec name (e.g. "GSM/8000/1").
	* \returns 0 on success, -1 on failure. */
	int mgcp_set_audio_info(void ctx, struct mgcp_rtp_codec codec,
	int payload_type, const char *audio_name)
	{
	int rate = codec->rate;
	int channels = codec->channels;
	char audio_codec[64];

	talloc_free(codec->subtype_name);
	codec->subtype_name = NULL;
	talloc_free(codec->audio_name);
	codec->audio_name = NULL;

	if (payload_type != PTYPE_UNDEFINED)
	codec->payload_type = payload_type;

	if (!audio_name) {
	switch (payload_type) {
	case 0:
	audio_name = "PCMU/8000/1";
	break;
	case 3:
	audio_name = "GSM/8000/1";
	break;
	case 8:
	audio_name = "PCMA/8000/1";
	break;
	case 18:
	audio_name = "G729/8000/1";
	break;
	default:
	/* Payload type is unknown, don't change rate and
	* channels. */
	/* TODO: return value? */
	return 0;
	}
	}

	if (sscanf(audio_name, "%63[^/]/%d/%d",
	audio_codec, &rate, &channels) < 1)
	return -EINVAL;

	codec->rate = rate;
	codec->channels = channels;
	codec->subtype_name = talloc_strdup(ctx, audio_codec);
	codec->audio_name = talloc_strdup(ctx, audio_name);

	if (!strcmp(audio_codec, "G729")) {
	codec->frame_duration_num = 10;
	codec->frame_duration_den = 1000;
	} else {
	codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
	codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
	}

	if (payload_type < 0) {
	payload_type = 96;
	if (rate == 8000 && channels == 1) {
	if (!strcmp(audio_codec, "GSM"))
	payload_type = 3;
	else if (!strcmp(audio_codec, "PCMA"))
	payload_type = 8;
	else if (!strcmp(audio_codec, "PCMU"))
	payload_type = 0;
	else if (!strcmp(audio_codec, "G729"))
	payload_type = 18;
	}

	codec->payload_type = payload_type;
	}

	if (channels != 1)
	LOGP(DLMGCP, LOGL_NOTICE,
	"Channels != 1 in SDP: '%s'\n", audio_name);

	return 0;
	}

	static void codecs_initialize(void ctx, struct sdp_rtp_map codecs, int used)
	{
	int i;

	for (i = 0; i < used; ++i) {
	switch (codecs[i].payload_type) {
	case 0:
	codecs[i].codec_name = "PCMU";
	codecs[i].rate = 8000;
	codecs[i].channels = 1;
	break;
	case 3:
	codecs[i].codec_name = "GSM";
	codecs[i].rate = 8000;
	codecs[i].channels = 1;
	break;
	case 8:
	codecs[i].codec_name = "PCMA";
	codecs[i].rate = 8000;
	codecs[i].channels = 1;
	break;
	case 18:
	codecs[i].codec_name = "G729";
	codecs[i].rate = 8000;
	codecs[i].channels = 1;
	break;
	}
	}
	}

	static void codecs_update(void ctx, struct sdp_rtp_map codecs, int used,
	int payload, const char *audio_name)
	{
	int i;

	for (i = 0; i < used; ++i) {
	char audio_codec[64];
	int rate = -1;
	int channels = -1;
	if (codecs[i].payload_type != payload)
	continue;
	if (sscanf(audio_name, "%63[^/]/%d/%d",
	audio_codec, &rate, &channels) < 1) {
	LOGP(DLMGCP, LOGL_ERROR, "Failed to parse '%s'\n",
	audio_name);
	continue;
	}

	codecs[i].map_line = talloc_strdup(ctx, audio_name);
	codecs[i].codec_name = talloc_strdup(ctx, audio_codec);
	codecs[i].rate = rate;
	codecs[i].channels = channels;
	return;
	}

	LOGP(DLMGCP, LOGL_ERROR, "Unconfigured PT(%d) with %s\n", payload,
	audio_name);
	}

	/* Check if the codec matches what is set up in the trunk config */
	static int is_codec_compatible(const struct mgcp_endpoint *endp,
	const struct sdp_rtp_map *codec)
	{
	char *codec_str;
	char audio_codec[64];

	if (!codec->codec_name)
	return 0;

	/* GSM, GSM/8000 and GSM/8000/1 should all be compatible...
	* let's go by name first. */
	codec_str = endp->tcfg->audio_name;
	if (sscanf(codec_str, "%63[^/]/%d/%d", audio_codec) < 1)
	return 0;

	return strcasecmp(audio_codec, codec->codec_name) == 0;
	}

	/*! Analyze SDP input string.
	* \param[in] endp trunk endpoint.
	* \param[out] conn associated rtp connection.
	* \param[out] caller provided memory to store the parsing results.
	* \returns 0 on success, -1 on failure.
	*
	* Note: In conn (conn->end) the function returns the packet duration,
	* the rtp port and the rtcp port */
	int mgcp_parse_sdp_data(const struct mgcp_endpoint *endp,
	struct mgcp_conn_rtp *conn,
	struct mgcp_parse_data *p)
	{
	struct sdp_rtp_map codecs[10];
	int codecs_used = 0;
	char *line;
	int maxptime = -1;
	int i;
	int codecs_assigned = 0;
	void *tmp_ctx = talloc_new(NULL);
	struct mgcp_rtp_end *rtp;

	int payload;
	int ptime, ptime2 = 0;
	char audio_name[64];
	int port, rc;
	char ipv4[16];

	OSMO_ASSERT(endp);
	OSMO_ASSERT(conn);
	OSMO_ASSERT(p);

	rtp = &conn->end;
	memset(&codecs, 0, sizeof(codecs));

	for_each_line(line, p->save) {
	switch (line[0]) {
	case 'o':
	case 's':
	case 't':
	case 'v':
	/* skip these SDP attributes */
	break;
	case 'a':
	if (sscanf(line, "a=rtpmap:%d %63s",
	&payload, audio_name) == 2) {
	codecs_update(tmp_ctx, codecs,
	codecs_used, payload, audio_name);
	} else
	if (sscanf
	(line, "a=ptime:%d-%d", &ptime, &ptime2) >= 1) {
	if (ptime2 > 0 && ptime2 != ptime)
	rtp->packet_duration_ms = 0;
	else
	rtp->packet_duration_ms = ptime;
	} else if (sscanf(line, "a=maxptime:%d", &ptime2)
	== 1) {
	maxptime = ptime2;
	}
	break;
	case 'm':
	rc = sscanf(line,
	"m=audio %d RTP/AVP %d %d %d %d %d %d %d %d %d %d",
	&port, &codecs[0].payload_type,
	&codecs[1].payload_type,
	&codecs[2].payload_type,
	&codecs[3].payload_type,
	&codecs[4].payload_type,
	&codecs[5].payload_type,
	&codecs[6].payload_type,
	&codecs[7].payload_type,
	&codecs[8].payload_type,
	&codecs[9].payload_type);
	if (rc >= 2) {
	rtp->rtp_port = htons(port);
	rtp->rtcp_port = htons(port + 1);
	codecs_used = rc - 1;
	codecs_initialize(tmp_ctx, codecs, codecs_used);
	}
	break;
	case 'c':

	if (sscanf(line, "c=IN IP4 %15s", ipv4) == 1) {
	inet_aton(ipv4, &rtp->addr);
	}
	break;
	default:
	if (p->endp)
	LOGP(DLMGCP, LOGL_NOTICE,
	"Unhandled SDP option: '%c'/%d on 0x%x\n",
	line[0], line[0],
	ENDPOINT_NUMBER(p->endp));
	else
	LOGP(DLMGCP, LOGL_NOTICE,
	"Unhandled SDP option: '%c'/%d\n",
	line[0], line[0]);
	break;
	}
	}

	/* Now select the primary and alt_codec */
	for (i = 0; i < codecs_used && codecs_assigned < 2; ++i) {
	struct mgcp_rtp_codec *codec = codecs_assigned == 0 ?
	&rtp->codec : &rtp->alt_codec;

	if (endp->tcfg->no_audio_transcoding &&
	!is_codec_compatible(endp, &codecs[i])) {
	LOGP(DLMGCP, LOGL_NOTICE, "Skipping codec %s\n",
	codecs[i].codec_name);
	continue;
	}

	mgcp_set_audio_info(p->cfg, codec,
	codecs[i].payload_type, codecs[i].map_line);
	codecs_assigned += 1;
	}

	if (codecs_assigned > 0) {
	/* TODO/XXX: Store this per codec and derive it on use */
	if (maxptime >= 0 && maxptime * rtp->codec.frame_duration_den >
	rtp->codec.frame_duration_num * 1500) {
	/* more than 1 frame */
	rtp->packet_duration_ms = 0;
	}

	LOGP(DLMGCP, LOGL_NOTICE,
	"Got media info via SDP: port %d, payload %d (%s), "
	"duration %d, addr %s\n",
	ntohs(rtp->rtp_port), rtp->codec.payload_type,
	rtp->codec.subtype_name ? rtp->
	codec.subtype_name : "unknown", rtp->packet_duration_ms,
	inet_ntoa(rtp->addr));
	}

	talloc_free(tmp_ctx);
	return codecs_assigned > 0;
	}

	/*! Generate SDP response string.
	* \param[in] endp trunk endpoint.
	* \param[in] conn associated rtp connection.
	* \param[out] sdp msg buffer to append resulting SDP string data.
	* \param[in] addr IPV4 address string (e.g. 192.168.100.1).
	* \returns 0 on success, -1 on failure. */
	int mgcp_write_response_sdp(const struct mgcp_endpoint *endp,
	const struct mgcp_conn_rtp conn, struct msgb sdp,
	const char *addr)
	{
	const char *fmtp_extra;
	const char *audio_name;
	int payload_type;
	int rc;

	OSMO_ASSERT(endp);
	OSMO_ASSERT(conn);
	OSMO_ASSERT(sdp);
	OSMO_ASSERT(addr);

	/* FIXME: constify endp and conn args in get_net_donwlink_format_cb() */
	endp->cfg->get_net_downlink_format_cb((struct mgcp_endpoint *)endp,
	&payload_type, &audio_name,
	&fmtp_extra,
	(struct mgcp_conn_rtp *)conn);

	rc = msgb_printf(sdp,
	"v=0\r\n"
	"o=- %s 23 IN IP4 %s\r\n"
	"s=-\r\n"
	"c=IN IP4 %s\r\n"
	"t=0 0\r\n", conn->conn->id, addr, addr);

	if (rc < 0)
	goto buffer_too_small;

	if (payload_type >= 0) {
	rc = msgb_printf(sdp, "m=audio %d RTP/AVP %d\r\n",
	conn->end.local_port, payload_type);
	if (rc < 0)
	goto buffer_too_small;

	if (audio_name && endp->tcfg->audio_send_name) {
	rc = msgb_printf(sdp, "a=rtpmap:%d %s\r\n",
	payload_type, audio_name);

	if (rc < 0)
	goto buffer_too_small;
	}

	if (fmtp_extra) {
	rc = msgb_printf(sdp, "%s\r\n", fmtp_extra);

	if (rc < 0)
	goto buffer_too_small;
	}
	}
	if (conn->end.packet_duration_ms > 0 && endp->tcfg->audio_send_ptime) {
	rc = msgb_printf(sdp, "a=ptime:%u\r\n",
	conn->end.packet_duration_ms);
	if (rc < 0)
	goto buffer_too_small;
	}

	return 0;

	buffer_too_small:
	LOGP(DLMGCP, LOGL_ERROR, "SDP messagebuffer too small\n");
	return -1;
	}