Merge branch 'zecke/features/sdp-codec-handling'
Move forward while preserving the legacy handling. Beging to
extract SDP rtpmap information and select codecs atfer this.
It is a foundation we can now build further and better check
ons.
diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h
index 9caab0b..485a124 100644
--- a/openbsc/include/openbsc/mgcp_internal.h
+++ b/openbsc/include/openbsc/mgcp_internal.h
@@ -22,6 +22,8 @@
#pragma once
+#include <string.h>
+
#include <osmocom/core/select.h>
#define CI_UNUSED 0
@@ -203,11 +205,51 @@
} osmux;
};
+#define for_each_line(line, save) \
+ for (line = strline_r(NULL, &save); line;\
+ line = strline_r(NULL, &save))
+
+static inline char *strline_r(char *str, char **saveptr)
+{
+ char *result;
+
+ if (str)
+ *saveptr = str;
+
+ result = *saveptr;
+
+ if (*saveptr != NULL) {
+ *saveptr = strpbrk(*saveptr, "\r\n");
+
+ if (*saveptr != NULL) {
+ char *eos = *saveptr;
+
+ if ((*saveptr)[0] == '\r' && (*saveptr)[1] == '\n')
+ (*saveptr)++;
+ (*saveptr)++;
+ if ((*saveptr)[0] == '\0')
+ *saveptr = NULL;
+
+ *eos = '\0';
+ }
+ }
+
+ return result;
+}
+
+
+
#define ENDPOINT_NUMBER(endp) abs((int)(endp - endp->tcfg->endpoints))
-struct mgcp_msg_ptr {
- unsigned int start;
- unsigned int length;
+/**
+ * Internal structure while parsing a request
+ */
+struct mgcp_parse_data {
+ struct mgcp_config *cfg;
+ struct mgcp_endpoint *endp;
+ char *trans;
+ char *save;
+ int found;
};
int mgcp_send_dummy(struct mgcp_endpoint *endp);
@@ -260,5 +302,21 @@
MGCP_DEST_BTS,
};
+
#define MGCP_DUMMY_LOAD 0x23
+
+/**
+ * SDP related information
+ */
+/* Assume audio frame length of 20ms */
+#define DEFAULT_RTP_AUDIO_FRAME_DUR_NUM 20
+#define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000
+#define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20
+#define DEFAULT_RTP_AUDIO_DEFAULT_RATE 8000
+#define DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS 1
+
+#define PTYPE_UNDEFINED (-1)
+int mgcp_parse_sdp_data(struct mgcp_endpoint *endp, struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p);
+int mgcp_set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
+ int payload_type, const char *audio_name);
diff --git a/openbsc/src/libmgcp/Makefile.am b/openbsc/src/libmgcp/Makefile.am
index d02b880..4403d60 100644
--- a/openbsc/src/libmgcp/Makefile.am
+++ b/openbsc/src/libmgcp/Makefile.am
@@ -8,7 +8,8 @@
noinst_HEADERS = g711common.h
-libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c mgcp_osmux.c
+libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c mgcp_osmux.c \
+ mgcp_sdp.c
if BUILD_MGCP_TRANSCODING
libmgcp_a_SOURCES += mgcp_transcode.c
diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c
index 62f6974..40ea791 100644
--- a/openbsc/src/libmgcp/mgcp_protocol.c
+++ b/openbsc/src/libmgcp/mgcp_protocol.c
@@ -24,7 +24,6 @@
#include <ctype.h>
#include <stdio.h>
#include <stdlib.h>
-#include <string.h>
#include <time.h>
#include <limits.h>
#include <unistd.h>
@@ -41,57 +40,9 @@
for (line = strtok_r(NULL, "\r\n", &save); line;\
line = strtok_r(NULL, "\r\n", &save))
-#define for_each_line(line, save) \
- for (line = strline_r(NULL, &save); line;\
- line = strline_r(NULL, &save))
-
-char *strline_r(char *str, char **saveptr)
-{
- char *result;
-
- if (str)
- *saveptr = str;
-
- result = *saveptr;
-
- if (*saveptr != NULL) {
- *saveptr = strpbrk(*saveptr, "\r\n");
-
- if (*saveptr != NULL) {
- char *eos = *saveptr;
-
- if ((*saveptr)[0] == '\r' && (*saveptr)[1] == '\n')
- (*saveptr)++;
- (*saveptr)++;
- if ((*saveptr)[0] == '\0')
- *saveptr = NULL;
-
- *eos = '\0';
- }
- }
-
- return result;
-}
-
-/* Assume audio frame length of 20ms */
-#define DEFAULT_RTP_AUDIO_FRAME_DUR_NUM 20
-#define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000
-#define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20
-#define DEFAULT_RTP_AUDIO_DEFAULT_RATE 8000
-#define DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS 1
-
-#define PTYPE_UNDEFINED (-1)
static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end);
-struct mgcp_parse_data {
- struct mgcp_config *cfg;
- struct mgcp_endpoint *endp;
- char *trans;
- char *save;
- int found;
-};
-
struct mgcp_request {
char *name;
struct msgb *(*handle_request) (struct mgcp_parse_data *data);
@@ -599,72 +550,6 @@
return ret;
}
-static int 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 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, "G729"))
- payload_type = 18;
- }
-
- codec->payload_type = payload_type;
- }
-
- if (channels != 1)
- LOGP(DMGCP, LOGL_NOTICE,
- "Channels != 1 in SDP: '%s'\n", audio_name);
-
- return 0;
-}
-
static int allocate_port(struct mgcp_endpoint *endp, struct mgcp_rtp_end *end,
struct mgcp_port_range *range,
int (*alloc)(struct mgcp_endpoint *endp, int port))
@@ -735,103 +620,6 @@
return 0;
}
-static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
-{
- char *line;
- int found_media = 0;
- /* TODO/XXX make it more generic */
- int audio_payload = -1;
- int audio_payload_alt = -1;
-
- for_each_line(line, p->save) {
- switch (line[0]) {
- case 'o':
- case 's':
- case 't':
- case 'v':
- /* skip these SDP attributes */
- break;
- case 'a': {
- int payload;
- int ptime, ptime2 = 0;
- char audio_name[64];
-
- if (audio_payload == -1)
- break;
-
- if (sscanf(line, "a=rtpmap:%d %63s",
- &payload, audio_name) == 2) {
- if (payload == audio_payload)
- set_audio_info(p->cfg, &rtp->codec,
- payload, audio_name);
- else if (payload == audio_payload_alt)
- set_audio_info(p->cfg, &rtp->alt_codec,
- 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) {
- /* TODO/XXX: Store this per codec and derive it on use */
- if (ptime2 * rtp->codec.frame_duration_den >
- rtp->codec.frame_duration_num * 1500)
- /* more than 1 frame */
- rtp->packet_duration_ms = 0;
- }
- break;
- }
- case 'm': {
- int port, rc;
- audio_payload = -1;
- audio_payload_alt = -1;
-
- rc = sscanf(line, "m=audio %d RTP/AVP %d %d",
- &port, &audio_payload, &audio_payload_alt);
- if (rc >= 2) {
- rtp->rtp_port = htons(port);
- rtp->rtcp_port = htons(port + 1);
- found_media = 1;
- set_audio_info(p->cfg, &rtp->codec, audio_payload, NULL);
- if (rc == 3)
- set_audio_info(p->cfg, &rtp->alt_codec,
- audio_payload_alt, NULL);
- }
- break;
- }
- case 'c': {
- char ipv4[16];
-
- if (sscanf(line, "c=IN IP4 %15s", ipv4) == 1) {
- inet_aton(ipv4, &rtp->addr);
- }
- break;
- }
- default:
- if (p->endp)
- LOGP(DMGCP, LOGL_NOTICE,
- "Unhandled SDP option: '%c'/%d on 0x%x\n",
- line[0], line[0], ENDPOINT_NUMBER(p->endp));
- else
- LOGP(DMGCP, LOGL_NOTICE,
- "Unhandled SDP option: '%c'/%d\n",
- line[0], line[0]);
- break;
- }
- }
-
- if (found_media)
- LOGP(DMGCP, 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));
-
- return found_media;
-}
-
/* Set the LCO from a string (see RFC 3435).
* The string is stored in the 'string' field. A NULL string is handled excatly
* like an empty string, the 'string' field is never NULL after this function
@@ -1036,13 +824,13 @@
endp->allocated = 1;
/* set up RTP media parameters */
- set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
+ mgcp_set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints,
tcfg->audio_fmtp_extra);
if (have_sdp)
- parse_sdp_data(&endp->net_end, p);
+ mgcp_parse_sdp_data(endp, &endp->net_end, p);
else if (endp->local_options.codec)
- set_audio_info(p->cfg, &endp->net_end.codec,
+ mgcp_set_audio_info(p->cfg, &endp->net_end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
if (p->cfg->bts_force_ptime) {
@@ -1143,7 +931,7 @@
case '\0':
/* SDP file begins */
have_sdp = 1;
- parse_sdp_data(&endp->net_end, p);
+ mgcp_parse_sdp_data(endp, &endp->net_end, p);
/* This will exhaust p->save, so the loop will
* terminate next time.
*/
@@ -1159,7 +947,7 @@
local_options);
if (!have_sdp && endp->local_options.codec)
- set_audio_info(p->cfg, &endp->net_end.codec,
+ mgcp_set_audio_info(p->cfg, &endp->net_end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
if (setup_rtp_processing(endp) != 0)
diff --git a/openbsc/src/libmgcp/mgcp_sdp.c b/openbsc/src/libmgcp/mgcp_sdp.c
new file mode 100644
index 0000000..33837b9
--- /dev/null
+++ b/openbsc/src/libmgcp/mgcp_sdp.c
@@ -0,0 +1,294 @@
+/*
+ * 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 <openbsc/mgcp.h>
+#include <openbsc/mgcp_internal.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;
+};
+
+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 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, "G729"))
+ payload_type = 18;
+ }
+
+ codec->payload_type = payload_type;
+ }
+
+ if (channels != 1)
+ LOGP(DMGCP, LOGL_NOTICE,
+ "Channels != 1 in SDP: '%s'\n", audio_name);
+
+ return 0;
+}
+
+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 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;
+ }
+ }
+}
+
+void codecs_update(void *ctx, struct sdp_rtp_map *codecs, int used, int payload, 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(DMGCP, 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(DMGCP, LOGL_ERROR, "Unconfigured PT(%d) with %s\n", payload, audio_name);
+}
+
+int is_codec_compatible(struct mgcp_endpoint *endp, struct sdp_rtp_map *codec)
+{
+ char *bts_codec;
+ char audio_codec[64];
+
+ /*
+ * GSM, GSM/8000 and GSM/8000/1 should all be compatible.. let's go
+ * by name first.
+ */
+ bts_codec = endp->tcfg->audio_name;
+ if (sscanf(bts_codec, "%63[^/]/%*d/%*d", audio_codec) < 1)
+ return 0;
+
+ return strcasecmp(audio_codec, codec->codec_name) == 0;
+}
+
+int mgcp_parse_sdp_data(struct mgcp_endpoint *endp, struct mgcp_rtp_end *rtp, 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);
+
+ 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': {
+ int payload;
+ int ptime, ptime2 = 0;
+ char audio_name[64];
+
+
+ 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': {
+ int port, rc;
+
+ 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': {
+ char ipv4[16];
+
+ if (sscanf(line, "c=IN IP4 %15s", ipv4) == 1) {
+ inet_aton(ipv4, &rtp->addr);
+ }
+ break;
+ }
+ default:
+ if (p->endp)
+ LOGP(DMGCP, LOGL_NOTICE,
+ "Unhandled SDP option: '%c'/%d on 0x%x\n",
+ line[0], line[0], ENDPOINT_NUMBER(p->endp));
+ else
+ LOGP(DMGCP, 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(DMGCP, 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(DMGCP, 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;
+}
+
diff --git a/openbsc/tests/mgcp/mgcp_test.c b/openbsc/tests/mgcp/mgcp_test.c
index 0f0e06c..d501859 100644
--- a/openbsc/tests/mgcp/mgcp_test.c
+++ b/openbsc/tests/mgcp/mgcp_test.c
@@ -340,6 +340,31 @@
"a=rtpmap:101 FOO/8000\r\n" \
"a=ptime:40\r\n"
+#define CRCX_MULT_GSM_EXACT \
+ "CRCX 259260421 5@mgw MGCP 1.0\r\n" \
+ "C: 1355c6041e\r\n" \
+ "I: 3\r\n" \
+ "L: p:20, a:GSM, nt:IN\r\n" \
+ "M: recvonly\r\n" \
+ "\r\n" \
+ "v=0\r\n" \
+ "o=- 1439038275 1439038275 IN IP4 192.168.181.247\r\n" \
+ "s=-\r\nc=IN IP4 192.168.181.247\r\n" \
+ "t=0 0\r\nm=audio 29084 RTP/AVP 0 8 3 18 4 96 97 101\r\n" \
+ "a=rtpmap:0 PCMU/8000\r\n" \
+ "a=rtpmap:8 PCMA/8000\r\n" \
+ "a=rtpmap:3 gsm/8000\r\n" \
+ "a=rtpmap:18 G729/8000\r\n" \
+ "a=fmtp:18 annexb=no\r\n" \
+ "a=rtpmap:4 G723/8000\r\n" \
+ "a=rtpmap:96 iLBC/8000\r\n" \
+ "a=fmtp:96 mode=20\r\n" \
+ "a=rtpmap:97 iLBC/8000\r\n" \
+ "a=fmtp:97 mode=30\r\n" \
+ "a=rtpmap:101 telephone-event/8000\r\n" \
+ "a=fmtp:101 0-15\r\n" \
+ "a=recvonly\r\n"
+
struct mgcp_test {
const char *name;
const char *req;
@@ -1011,6 +1036,40 @@
OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
OSMO_ASSERT(endp->net_end.alt_codec.payload_type == -1);
+ /* Allocate 5@mgw at select GSM.. */
+ last_endpoint = -1;
+ inp = create_msg(CRCX_MULT_GSM_EXACT);
+ talloc_free(cfg->trunk.audio_name);
+ cfg->trunk.audio_name = "GSM/8000";
+ cfg->trunk.no_audio_transcoding = 1;
+ resp = mgcp_handle_message(cfg, inp);
+ msgb_free(inp);
+ msgb_free(resp);
+
+ OSMO_ASSERT(last_endpoint == 5);
+ endp = &cfg->trunk.endpoints[last_endpoint];
+ OSMO_ASSERT(endp->net_end.codec.payload_type == 3);
+ OSMO_ASSERT(endp->net_end.alt_codec.payload_type == -1);
+
+ /* Check what happens without that flag */
+
+ /* Free the previous endpoint and the data ... */
+ mgcp_release_endp(endp);
+ talloc_free(endp->last_response);
+ talloc_free(endp->last_trans);
+ endp->last_response = endp->last_trans = NULL;
+
+ last_endpoint = -1;
+ inp = create_msg(CRCX_MULT_GSM_EXACT);
+ cfg->trunk.no_audio_transcoding = 0;
+ resp = mgcp_handle_message(cfg, inp);
+ msgb_free(inp);
+ msgb_free(resp);
+
+ OSMO_ASSERT(last_endpoint == 5);
+ endp = &cfg->trunk.endpoints[last_endpoint];
+ OSMO_ASSERT(endp->net_end.codec.payload_type == 0);
+ OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 8);
talloc_free(cfg);
}