Hello Harald Welte, Jenkins Builder,
I'd like you to reexamine a change. Please visit
https://gerrit.osmocom.org/4006
to look at the new patch set (#6).
sdp: refactoring sdp parser/generator
move SDP generator function write_response_sdp() from mgcp_protocol.c
to mgcp_sdp.c
move prototypes for mgcp_parse_sdp_data() and mgcp_set_audio_info()
to mgcp_sdp.h
change parameter list of mgcp_parse_sdp_data() so that it takes the
rtp conn directly, rather than struct mgcp_rtp_end.
fix sourcecode formatting in mgcp_sdp.c
add doxygen comments to all public functions
Change-Id: I9f88c93872ff913bc211f560b26901267f577324
---
M include/osmocom/mgcp/Makefile.am
M include/osmocom/mgcp/mgcp_internal.h
A include/osmocom/mgcp/mgcp_sdp.h
M src/libosmo-mgcp/mgcp_protocol.c
M src/libosmo-mgcp/mgcp_sdp.c
5 files changed, 211 insertions(+), 142 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-mgw refs/changes/06/4006/6
diff --git a/include/osmocom/mgcp/Makefile.am b/include/osmocom/mgcp/Makefile.am
index 646b887..cd8f599 100644
--- a/include/osmocom/mgcp/Makefile.am
+++ b/include/osmocom/mgcp/Makefile.am
@@ -4,4 +4,5 @@
mgcp_conn.h \
mgcp_stat.h \
mgcp_ep.h \
+ mgcp_sdp.h \
$(NULL)
diff --git a/include/osmocom/mgcp/mgcp_internal.h
b/include/osmocom/mgcp/mgcp_internal.h
index 3a22d51..777d787 100644
--- a/include/osmocom/mgcp/mgcp_internal.h
+++ b/include/osmocom/mgcp/mgcp_internal.h
@@ -317,9 +317,6 @@
#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);
/*! get the ip-address where the mgw application is bound on.
* \param[in] endp mgcp endpoint, that holds a copy of the VTY parameters
diff --git a/include/osmocom/mgcp/mgcp_sdp.h b/include/osmocom/mgcp/mgcp_sdp.h
new file mode 100644
index 0000000..0eb376d
--- /dev/null
+++ b/include/osmocom/mgcp/mgcp_sdp.h
@@ -0,0 +1,33 @@
+/*
+ * SDP generation and parsing
+ *
+ * (C) 2009-2015 by Holger Hans Peter Freyther <[email protected]>
+ * (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/>.
+ *
+ */
+
+#pragma once
+
+int mgcp_parse_sdp_data(struct mgcp_endpoint *endp, struct mgcp_conn_rtp *conn,
+ struct mgcp_parse_data *p);
+
+int mgcp_set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
+ int payload_type, const char *audio_name);
+
+int mgcp_write_response_sdp(struct mgcp_endpoint *endp,
+ struct mgcp_conn_rtp *conn, char *sdp_record,
+ size_t size, const char *addr);
diff --git a/src/libosmo-mgcp/mgcp_protocol.c b/src/libosmo-mgcp/mgcp_protocol.c
index ac25894..5d4dd47 100644
--- a/src/libosmo-mgcp/mgcp_protocol.c
+++ b/src/libosmo-mgcp/mgcp_protocol.c
@@ -38,6 +38,7 @@
#include <osmocom/mgcp/mgcp_stat.h>
#include <osmocom/mgcp/mgcp_msg.h>
#include <osmocom/mgcp/mgcp_ep.h>
+#include <osmocom/mgcp/mgcp_sdp.h>
struct mgcp_request {
char *name;
@@ -190,80 +191,6 @@
return create_resp(endp, code, " FAIL", msg, trans, NULL, NULL);
}
-static int write_response_sdp(struct mgcp_endpoint *endp,
- struct mgcp_conn_rtp *conn,
- char *sdp_record, size_t size, const char *addr)
-{
- const char *fmtp_extra;
- const char *audio_name;
- int payload_type;
- int len;
- int nchars;
-
- if (!conn)
- return -1;
-
- endp->cfg->get_net_downlink_format_cb(endp, &payload_type,
- &audio_name, &fmtp_extra, conn);
-
- len = snprintf(sdp_record, size,
- "v=0\r\n"
- "o=- %u 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 (len < 0 || len >= size)
- goto buffer_too_small;
-
- if (payload_type >= 0) {
- nchars = snprintf(sdp_record + len, size - len,
- "m=audio %d RTP/AVP %d\r\n",
- conn->end.local_port, payload_type);
- if (nchars < 0 || nchars >= size - len)
- goto buffer_too_small;
-
- len += nchars;
-
- if (audio_name && endp->tcfg->audio_send_name) {
- nchars = snprintf(sdp_record + len, size - len,
- "a=rtpmap:%d %s\r\n",
- payload_type, audio_name);
-
- if (nchars < 0 || nchars >= size - len)
- goto buffer_too_small;
-
- len += nchars;
- }
-
- if (fmtp_extra) {
- nchars = snprintf(sdp_record + len, size - len,
- "%s\r\n", fmtp_extra);
-
- if (nchars < 0 || nchars >= size - len)
- goto buffer_too_small;
-
- len += nchars;
- }
- }
- if (conn->end.packet_duration_ms > 0 && endp->tcfg->audio_send_ptime) {
- nchars = snprintf(sdp_record + len, size - len,
- "a=ptime:%u\r\n",
- conn->end.packet_duration_ms);
- if (nchars < 0 || nchars >= size - len)
- goto buffer_too_small;
-
- len += nchars;
- }
-
- return len;
-
-buffer_too_small:
- LOGP(DLMGCP, LOGL_ERROR, "SDP buffer too small: %zu (needed %d)\n",
- size, len);
- return -1;
-}
-
/* Format MGCP response string (with SDP attached) */
static struct msgb *create_response_with_sdp(struct mgcp_endpoint *endp,
struct mgcp_conn_rtp *conn,
@@ -291,8 +218,8 @@
if (len < 0)
return NULL;
- nchars = write_response_sdp(endp, conn, sdp_record + len,
- sizeof(sdp_record) - len - 1, addr);
+ nchars = mgcp_write_response_sdp(endp, conn, sdp_record + len,
+ sizeof(sdp_record) - len - 1, addr);
if (nchars < 0)
return NULL;
@@ -688,7 +615,7 @@
/* set up RTP media parameters */
if (have_sdp)
- mgcp_parse_sdp_data(endp, &conn->end, p);
+ mgcp_parse_sdp_data(endp, conn, p);
else if (endp->local_options.codec)
mgcp_set_audio_info(p->cfg, &conn->end.codec,
PTYPE_UNDEFINED, endp->local_options.codec);
@@ -835,7 +762,7 @@
conn->conn->mode = conn->conn->mode_orig;
if (have_sdp)
- mgcp_parse_sdp_data(endp, &conn->end, p);
+ mgcp_parse_sdp_data(endp, conn, p);
set_local_cx_options(endp->tcfg->endpoints, &endp->local_options,
local_options);
diff --git a/src/libosmo-mgcp/mgcp_sdp.c b/src/libosmo-mgcp/mgcp_sdp.c
index 7568351..048a533 100644
--- a/src/libosmo-mgcp/mgcp_sdp.c
+++ b/src/libosmo-mgcp/mgcp_sdp.c
@@ -38,6 +38,12 @@
int channels;
};
+/*! set codec configuration depending on payload type and codec name.
+ * \endp[in] ctx talloc context
+ * \endp[out] codec configuration (caller provided memory)
+ * \endp[in] payload_type codec type id (e.g. 3 for GSM, -1 when undefined)
+ * \endp[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)
{
@@ -55,15 +61,23 @@
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;
+ 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;
+ /* Payload type is unknown, don't change rate and
+ * channels. */
+ /* TODO: return value? */
+ return 0;
}
}
@@ -107,7 +121,7 @@
return 0;
}
-void codecs_initialize(void *ctx, struct sdp_rtp_map *codecs, int used)
+static void codecs_initialize(void *ctx, struct sdp_rtp_map *codecs, int used)
{
int i;
@@ -137,7 +151,8 @@
}
}
-void codecs_update(void *ctx, struct sdp_rtp_map *codecs, int used, int
payload, char *audio_name)
+static void codecs_update(void *ctx, struct sdp_rtp_map *codecs, int used,
+ int payload, char *audio_name)
{
int i;
@@ -148,8 +163,9 @@
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);
+ audio_codec, &rate, &channels) < 1) {
+ LOGP(DLMGCP, LOGL_ERROR, "Failed to parse '%s'\n",
+ audio_name);
continue;
}
@@ -160,29 +176,36 @@
return;
}
- LOGP(DLMGCP, LOGL_ERROR, "Unconfigured PT(%d) with %s\n", payload,
audio_name);
+ LOGP(DLMGCP, LOGL_ERROR, "Unconfigured PT(%d) with %s\n", payload,
+ audio_name);
}
-int is_codec_compatible(struct mgcp_endpoint *endp, struct sdp_rtp_map *codec)
+/* Check if the codec matches what is set up in the trunk config */
+static int is_codec_compatible(struct mgcp_endpoint *endp,
+ struct sdp_rtp_map *codec)
{
- char *bts_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.
- */
- bts_codec = endp->tcfg->audio_name;
- if (sscanf(bts_codec, "%63[^/]/%*d/%*d", audio_codec) < 1)
+ /* 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;
}
-int mgcp_parse_sdp_data(struct mgcp_endpoint *endp, struct mgcp_rtp_end *rtp,
struct mgcp_parse_data *p)
+/*! analyze SDP input string.
+ * \endp[in] endp trunk endpoint
+ * \endp[in] conn associated rtp connection
+ * \endp[in] caller provided memory to store the parsing results
+ * \returns 0 on success, -1 on failure */
+int mgcp_parse_sdp_data(struct mgcp_endpoint *endp, struct mgcp_conn_rtp *conn,
+ struct mgcp_parse_data *p)
{
struct sdp_rtp_map codecs[10];
int codecs_used = 0;
@@ -191,7 +214,19 @@
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) {
@@ -202,41 +237,36 @@
case 'v':
/* skip these SDP attributes */
break;
- case 'a': {
- int payload;
- int ptime, ptime2 = 0;
- char audio_name[64];
-
-
+ 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) {
+ 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)
{
+ } 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);
+ 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);
@@ -244,20 +274,18 @@
codecs_initialize(tmp_ctx, codecs, codecs_used);
}
break;
- }
- case 'c': {
- char ipv4[16];
+ 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));
+ line[0], line[0],
+ ENDPOINT_NUMBER(p->endp));
else
LOGP(DLMGCP, LOGL_NOTICE,
"Unhandled SDP option: '%c'/%d\n",
@@ -269,25 +297,24 @@
/* 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;
+ &rtp->codec : &rtp->alt_codec;
if (endp->tcfg->no_audio_transcoding &&
- !is_codec_compatible(endp, &codecs[i])) {
+ !is_codec_compatible(endp, &codecs[i])) {
LOGP(DLMGCP, LOGL_NOTICE, "Skipping codec %s\n",
- codecs[i].codec_name);
+ codecs[i].codec_name);
continue;
}
mgcp_set_audio_info(p->cfg, codec,
- codecs[i].payload_type,
- codecs[i].map_line);
+ 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) {
+ rtp->codec.frame_duration_num * 1500) {
/* more than 1 frame */
rtp->packet_duration_ms = 0;
}
@@ -296,11 +323,95 @@
"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));
+ 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.
+ * \endp[in] endp trunk endpoint
+ * \endp[in] conn associated rtp connection
+ * \endp[out] sdp_record resulting SDP string
+ * \endp[in] size buffer size of sdp_record
+ * \endp[in] addr IPV4 address string (e.g. 192.168.100.1)
+ * \returns 0 on success, -1 on failure */
+int mgcp_write_response_sdp(struct mgcp_endpoint *endp,
+ struct mgcp_conn_rtp *conn, char *sdp_record,
+ size_t size, const char *addr)
+{
+ const char *fmtp_extra;
+ const char *audio_name;
+ int payload_type;
+ int len;
+ int nchars;
+
+ OSMO_ASSERT(endp);
+ OSMO_ASSERT(conn);
+ OSMO_ASSERT(sdp_record);
+ OSMO_ASSERT(size > 0);
+ OSMO_ASSERT(addr);
+
+ endp->cfg->get_net_downlink_format_cb(endp, &payload_type,
+ &audio_name, &fmtp_extra, conn);
+
+ len = snprintf(sdp_record, size,
+ "v=0\r\n"
+ "o=- %u 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 (len < 0 || len >= size)
+ goto buffer_too_small;
+
+ if (payload_type >= 0) {
+ nchars = snprintf(sdp_record + len, size - len,
+ "m=audio %d RTP/AVP %d\r\n",
+ conn->end.local_port, payload_type);
+ if (nchars < 0 || nchars >= size - len)
+ goto buffer_too_small;
+
+ len += nchars;
+
+ if (audio_name && endp->tcfg->audio_send_name) {
+ nchars = snprintf(sdp_record + len, size - len,
+ "a=rtpmap:%d %s\r\n",
+ payload_type, audio_name);
+
+ if (nchars < 0 || nchars >= size - len)
+ goto buffer_too_small;
+
+ len += nchars;
+ }
+
+ if (fmtp_extra) {
+ nchars = snprintf(sdp_record + len, size - len,
+ "%s\r\n", fmtp_extra);
+
+ if (nchars < 0 || nchars >= size - len)
+ goto buffer_too_small;
+
+ len += nchars;
+ }
+ }
+ if (conn->end.packet_duration_ms > 0 && endp->tcfg->audio_send_ptime) {
+ nchars = snprintf(sdp_record + len, size - len,
+ "a=ptime:%u\r\n",
+ conn->end.packet_duration_ms);
+ if (nchars < 0 || nchars >= size - len)
+ goto buffer_too_small;
+
+ len += nchars;
+ }
+
+ return len;
+
+buffer_too_small:
+ LOGP(DLMGCP, LOGL_ERROR, "SDP buffer too small: %zu (needed %d)\n",
+ size, len);
+ return -1;
+}
--
To view, visit https://gerrit.osmocom.org/4006
To unsubscribe, visit https://gerrit.osmocom.org/settings
Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I9f88c93872ff913bc211f560b26901267f577324
Gerrit-PatchSet: 6
Gerrit-Project: osmo-mgw
Gerrit-Branch: master
Gerrit-Owner: Neels Hofmeyr <[email protected]>
Gerrit-Reviewer: Harald Welte <[email protected]>
Gerrit-Reviewer: Jenkins Builder
Gerrit-Reviewer: dexter <[email protected]>
