diff options
author | Holger Hans Peter Freyther <holger@moiji-mobile.com> | 2015-08-14 15:48:54 +0200 |
---|---|---|
committer | Holger Hans Peter Freyther <holger@moiji-mobile.com> | 2015-08-14 15:48:54 +0200 |
commit | a334e90ddf99697ad6b18df80f1cd7473b2314d4 (patch) | |
tree | f7a50b71838b0f868e9d8d4ad6871c4f3d4d9029 | |
parent | e9f7c9925c26b23f7c29ace8da381e439a658eeb (diff) | |
parent | aeadf261e54d4e3987797b5818a8356441512568 (diff) |
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.
-rw-r--r-- | openbsc/include/openbsc/mgcp_internal.h | 64 | ||||
-rw-r--r-- | openbsc/src/libmgcp/Makefile.am | 3 | ||||
-rw-r--r-- | openbsc/src/libmgcp/mgcp_protocol.c | 222 | ||||
-rw-r--r-- | openbsc/src/libmgcp/mgcp_sdp.c | 294 | ||||
-rw-r--r-- | openbsc/tests/mgcp/mgcp_test.c | 59 |
5 files changed, 421 insertions, 221 deletions
diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h index 9caab0b21..485a12409 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 @@ struct mgcp_endpoint { } 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 @@ enum { 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 d02b880e6..4403d6086 100644 --- a/openbsc/src/libmgcp/Makefile.am +++ b/openbsc/src/libmgcp/Makefile.am @@ -8,7 +8,8 @@ noinst_LIBRARIES = libmgcp.a 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 62f6974d0..40ea7916a 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 @@ static int parse_conn_mode(const char *msg, struct mgcp_endpoint *endp) 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 @@ static int allocate_ports(struct mgcp_endpoint *endp) 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 @@ mgcp_header_done: 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 @@ static struct msgb *handle_modify_con(struct mgcp_parse_data *p) 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 @@ static struct msgb *handle_modify_con(struct mgcp_parse_data *p) 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 000000000..33837b9af --- /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 0f0e06ccf..d5018591b 100644 --- a/openbsc/tests/mgcp/mgcp_test.c +++ b/openbsc/tests/mgcp/mgcp_test.c @@ -340,6 +340,31 @@ static void test_strline(void) "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 @@ static void test_multilple_codec(void) 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); } |