[asterisk-commits] SDP: Explicitly stop a RTP instance before destoying it. (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri May 12 12:29:40 CDT 2017
George Joseph has submitted this change and it was merged. ( https://gerrit.asterisk.org/5598 )
Change subject: SDP: Explicitly stop a RTP instance before destoying it.
......................................................................
SDP: Explicitly stop a RTP instance before destoying it.
* Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream()
handle generating disabled/declined streams.
* Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not
check the offerer side negotiated SDP because that isn't the purpose of
this patch and there is much to be done to handle declined/dummy streams.
* Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and
/main/sdp/sdp_merge_crisscross unit tests.
Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31
---
M include/asterisk/stream.h
M main/sdp_state.c
M tests/test_sdp.c
3 files changed, 438 insertions(+), 188 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved
Jenkins2: Approved for Submit
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/include/asterisk/stream.h b/include/asterisk/stream.h
index 526aee1..b453ab9 100644
--- a/include/asterisk/stream.h
+++ b/include/asterisk/stream.h
@@ -56,7 +56,7 @@
*/
enum ast_stream_state {
/*!
- * \brief Set when the stream has been removed
+ * \brief Set when the stream has been removed/declined
*/
AST_STREAM_STATE_REMOVED = 0,
/*!
diff --git a/main/sdp_state.c b/main/sdp_state.c
index 598a610..9b116ca 100644
--- a/main/sdp_state.c
+++ b/main/sdp_state.c
@@ -64,6 +64,11 @@
typedef int (*state_fn)(struct ast_sdp_state *state);
+struct sdp_state_rtp {
+ /*! The underlying RTP instance */
+ struct ast_rtp_instance *instance;
+};
+
struct sdp_state_udptl {
/*! The underlying UDPTL instance */
struct ast_udptl *instance;
@@ -74,7 +79,7 @@
enum ast_media_type type;
union {
/*! The underlying RTP instance */
- struct ast_rtp_instance *instance;
+ struct sdp_state_rtp *rtp;
/*! The underlying UDPTL instance */
struct sdp_state_udptl *udptl;
};
@@ -85,6 +90,16 @@
/*! UDPTL session parameters */
struct ast_control_t38_parameters t38_local_params;
};
+
+static void sdp_state_rtp_destroy(void *obj)
+{
+ struct sdp_state_rtp *rtp = obj;
+
+ if (rtp->instance) {
+ ast_rtp_instance_stop(rtp->instance);
+ ast_rtp_instance_destroy(rtp->instance);
+ }
+}
static void sdp_state_udptl_destroy(void *obj)
{
@@ -100,9 +115,7 @@
switch (state_stream->type) {
case AST_MEDIA_TYPE_AUDIO:
case AST_MEDIA_TYPE_VIDEO:
- if (state_stream->instance) {
- ast_rtp_instance_destroy(state_stream->instance);
- }
+ ao2_cleanup(state_stream->rtp);
break;
case AST_MEDIA_TYPE_IMAGE:
ao2_cleanup(state_stream->udptl);
@@ -145,10 +158,10 @@
static struct ast_sched_context *sched;
/*! \brief Internal function which creates an RTP instance */
-static struct ast_rtp_instance *create_rtp(const struct ast_sdp_options *options,
+static struct sdp_state_rtp *create_rtp(const struct ast_sdp_options *options,
enum ast_media_type media_type)
{
- struct ast_rtp_instance *rtp;
+ struct sdp_state_rtp *rtp;
struct ast_rtp_engine_ice *ice;
static struct ast_sockaddr address_rtp;
struct ast_sockaddr *media_address = &address_rtp;
@@ -167,38 +180,57 @@
}
}
- rtp = ast_rtp_instance_new(options->rtp_engine, sched, media_address, NULL);
+ rtp = ao2_alloc_options(sizeof(*rtp), sdp_state_rtp_destroy, AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!rtp) {
- ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n",
- options->rtp_engine);
return NULL;
}
- ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
- ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_NAT, options->rtp_symmetric);
+ rtp->instance = ast_rtp_instance_new(options->rtp_engine, sched, media_address, NULL);
+ if (!rtp->instance) {
+ ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n",
+ options->rtp_engine);
+ ao2_ref(rtp, -1);
+ return NULL;
+ }
- if (options->ice == AST_SDP_ICE_DISABLED && (ice = ast_rtp_instance_get_ice(rtp))) {
- ice->stop(rtp);
+ ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_RTCP,
+ AST_RTP_INSTANCE_RTCP_STANDARD);
+ ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_NAT,
+ options->rtp_symmetric);
+
+ if (options->ice == AST_SDP_ICE_DISABLED
+ && (ice = ast_rtp_instance_get_ice(rtp->instance))) {
+ ice->stop(rtp->instance);
}
if (options->dtmf == AST_SDP_DTMF_RFC_4733 || options->dtmf == AST_SDP_DTMF_AUTO) {
- ast_rtp_instance_dtmf_mode_set(rtp, AST_RTP_DTMF_MODE_RFC2833);
- ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_DTMF, 1);
+ ast_rtp_instance_dtmf_mode_set(rtp->instance, AST_RTP_DTMF_MODE_RFC2833);
+ ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_DTMF, 1);
} else if (options->dtmf == AST_SDP_DTMF_INBAND) {
- ast_rtp_instance_dtmf_mode_set(rtp, AST_RTP_DTMF_MODE_INBAND);
+ ast_rtp_instance_dtmf_mode_set(rtp->instance, AST_RTP_DTMF_MODE_INBAND);
}
- if (media_type == AST_MEDIA_TYPE_AUDIO &&
- (options->tos_audio || options->cos_audio)) {
- ast_rtp_instance_set_qos(rtp, options->tos_audio,
- options->cos_audio, "SIP RTP Audio");
- } else if (media_type == AST_MEDIA_TYPE_VIDEO &&
- (options->tos_video || options->cos_video)) {
- ast_rtp_instance_set_qos(rtp, options->tos_video,
- options->cos_video, "SIP RTP Video");
+ switch (media_type) {
+ case AST_MEDIA_TYPE_AUDIO:
+ if (options->tos_audio || options->cos_audio) {
+ ast_rtp_instance_set_qos(rtp->instance, options->tos_audio,
+ options->cos_audio, "SIP RTP Audio");
+ }
+ break;
+ case AST_MEDIA_TYPE_VIDEO:
+ if (options->tos_video || options->cos_video) {
+ ast_rtp_instance_set_qos(rtp->instance, options->tos_video,
+ options->cos_video, "SIP RTP Video");
+ }
+ break;
+ case AST_MEDIA_TYPE_IMAGE:
+ case AST_MEDIA_TYPE_TEXT:
+ case AST_MEDIA_TYPE_UNKNOWN:
+ case AST_MEDIA_TYPE_END:
+ break;
}
- ast_rtp_instance_set_last_rx(rtp, time(NULL));
+ ast_rtp_instance_set_last_rx(rtp->instance, time(NULL));
return rtp;
}
@@ -278,8 +310,8 @@
switch (state_stream->type) {
case AST_MEDIA_TYPE_AUDIO:
case AST_MEDIA_TYPE_VIDEO:
- state_stream->instance = create_rtp(options, state_stream->type);
- if (!state_stream->instance) {
+ state_stream->rtp = create_rtp(options, state_stream->type);
+ if (!state_stream->rtp) {
sdp_state_stream_free(state_stream);
sdp_state_capabilities_free(capabilities);
return NULL;
@@ -413,11 +445,11 @@
sdp_state->proposed_capabilities->topology, stream_index)) == AST_MEDIA_TYPE_VIDEO);
stream_state = sdp_state_get_stream(sdp_state, stream_index);
- if (!stream_state) {
+ if (!stream_state || !stream_state->rtp) {
return NULL;
}
- return stream_state->instance;
+ return stream_state->rtp->instance;
}
struct ast_udptl *ast_sdp_state_get_udptl_instance(
@@ -467,7 +499,7 @@
stream_index))) {
case AST_MEDIA_TYPE_AUDIO:
case AST_MEDIA_TYPE_VIDEO:
- ast_rtp_instance_get_local_address(stream_state->instance, address);
+ ast_rtp_instance_get_local_address(stream_state->rtp->instance, address);
break;
case AST_MEDIA_TYPE_IMAGE:
ast_udptl_get_us(stream_state->udptl->instance, address);
@@ -730,7 +762,7 @@
switch (joint_state_stream->type) {
case AST_MEDIA_TYPE_AUDIO:
case AST_MEDIA_TYPE_VIDEO:
- joint_state_stream->instance = ao2_bump(local_state_stream->instance);
+ joint_state_stream->rtp = ao2_bump(local_state_stream->rtp);
if (is_local) {
break;
}
@@ -744,8 +776,8 @@
ast_rtp_codecs_payloads_xover(codecs, codecs, NULL);
}
ast_rtp_codecs_payloads_copy(codecs,
- ast_rtp_instance_get_codecs(joint_state_stream->instance),
- joint_state_stream->instance);
+ ast_rtp_instance_get_codecs(joint_state_stream->rtp->instance),
+ joint_state_stream->rtp->instance);
break;
case AST_MEDIA_TYPE_IMAGE:
joint_state_stream->udptl = ao2_bump(local_state_stream->udptl);
@@ -777,9 +809,9 @@
switch (new_stream_type) {
case AST_MEDIA_TYPE_AUDIO:
case AST_MEDIA_TYPE_VIDEO:
- joint_state_stream->instance = create_rtp(sdp_state->options,
+ joint_state_stream->rtp = create_rtp(sdp_state->options,
new_stream_type);
- if (!joint_state_stream->instance) {
+ if (!joint_state_stream->rtp) {
ast_stream_free(joint_stream);
sdp_state_stream_free(joint_state_stream);
goto fail;
@@ -954,24 +986,33 @@
* sides.
*
* \param state The SDP state in which SDPs have been negotiated
- * \param rtp The RTP instance that is being updated
+ * \param rtp The RTP wrapper that is being updated
* \param options Our locally-supported SDP options
* \param remote_sdp The SDP we most recently received
* \param remote_m_line The remote SDP stream that corresponds to the RTP instance we are modifying
*/
-static void update_rtp_after_merge(const struct ast_sdp_state *state, struct ast_rtp_instance *rtp,
+static void update_rtp_after_merge(const struct ast_sdp_state *state,
+ struct sdp_state_rtp *rtp,
const struct ast_sdp_options *options,
const struct ast_sdp *remote_sdp,
const struct ast_sdp_m_line *remote_m_line)
{
- if (ast_sdp_options_get_rtcp_mux(options) && ast_sdp_m_find_attribute(remote_m_line, "rtcp-mux", -1)) {
- ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
+ if (!rtp) {
+ /* This is a dummy stream */
+ return;
+ }
+
+ if (ast_sdp_options_get_rtcp_mux(options)
+ && ast_sdp_m_find_attribute(remote_m_line, "rtcp-mux", -1)) {
+ ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_RTCP,
+ AST_RTP_INSTANCE_RTCP_MUX);
} else {
- ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
+ ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_RTCP,
+ AST_RTP_INSTANCE_RTCP_STANDARD);
}
if (ast_sdp_options_get_ice(options) == AST_SDP_ICE_ENABLED_STANDARD) {
- update_ice(state, rtp, options, remote_sdp, remote_m_line);
+ update_ice(state, rtp->instance, options, remote_sdp, remote_m_line);
}
}
@@ -999,6 +1040,11 @@
struct ast_sdp_c_line *c_line;
unsigned int fax_max_datagram;
struct ast_sockaddr *addrs;
+
+ if (!udptl) {
+ /* This is a dummy stream */
+ return;
+ }
a_line = ast_sdp_m_find_attribute(remote_m_line, "t38faxmaxdatagram", -1);
if (!a_line) {
@@ -1103,7 +1149,7 @@
switch (ast_stream_get_type(ast_stream_topology_get_stream(joint_capabilities->topology, i))) {
case AST_MEDIA_TYPE_AUDIO:
case AST_MEDIA_TYPE_VIDEO:
- update_rtp_after_merge(sdp_state, state_stream->instance, sdp_state->options,
+ update_rtp_after_merge(sdp_state, state_stream->rtp, sdp_state->options,
remote_sdp, ast_sdp_get_m(remote_sdp, i));
break;
case AST_MEDIA_TYPE_IMAGE:
@@ -1312,122 +1358,162 @@
struct ast_format_cap *caps;
int i;
int rtp_code;
+ int rtp_port;
int min_packet_size = 0;
int max_packet_size = 0;
enum ast_media_type media_type;
char tmp[64];
- struct ast_sockaddr address_rtp;
+ struct sdp_state_stream *stream_state;
struct ast_rtp_instance *rtp;
struct ast_sdp_a_line *a_line;
stream = ast_stream_topology_get_stream(capabilities->topology, stream_index);
- rtp = AST_VECTOR_GET(&capabilities->streams, stream_index)->instance;
ast_assert(sdp && options && stream);
+ caps = ast_stream_get_formats(stream);
+
+ stream_state = AST_VECTOR_GET(&capabilities->streams, stream_index);
+ if (stream_state->rtp && caps && ast_format_cap_count(caps)) {
+ rtp = stream_state->rtp->instance;
+ } else {
+ /* This is a disabled stream */
+ rtp = NULL;
+ }
+
if (rtp) {
+ struct ast_sockaddr address_rtp;
+
if (ast_sdp_state_get_stream_connection_address(sdp_state, 0, &address_rtp)) {
return -1;
}
+ rtp_port = ast_sockaddr_port(&address_rtp);
} else {
- ast_sockaddr_setnull(&address_rtp);
+ rtp_port = 0;
}
m_line = ast_sdp_m_alloc(
ast_codec_media_type2str(ast_stream_get_type(stream)),
- ast_sockaddr_port(&address_rtp), 1,
+ rtp_port, 1,
options->encryption != AST_SDP_ENCRYPTION_DISABLED ? "RTP/SAVP" : "RTP/AVP",
NULL);
if (!m_line) {
return -1;
}
- caps = ast_stream_get_formats(stream);
+ if (rtp_port) {
+ /* Stream is not declined/disabled */
+ for (i = 0; i < ast_format_cap_count(caps); i++) {
+ struct ast_format *format = ast_format_cap_get_format(caps, i);
- for (i = 0; i < ast_format_cap_count(caps); i++) {
- struct ast_format *format = ast_format_cap_get_format(caps, i);
+ rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp), 1,
+ format, 0);
+ if (rtp_code == -1) {
+ ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n",
+ ast_format_get_name(format));
+ ao2_ref(format, -1);
+ continue;
+ }
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp), 1, format, 0)) == -1) {
- ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
- ao2_ref(format, -1);
- continue;
- }
-
- if (ast_sdp_m_add_format(m_line, options, rtp_code, 1, format, 0)) {
- ast_sdp_m_free(m_line);
- ao2_ref(format, -1);
- return -1;
- }
-
- if (ast_format_get_maximum_ms(format) &&
- ((ast_format_get_maximum_ms(format) < max_packet_size) || !max_packet_size)) {
- max_packet_size = ast_format_get_maximum_ms(format);
- }
-
- ao2_ref(format, -1);
- }
-
- media_type = ast_stream_get_type(stream);
- if (rtp && media_type != AST_MEDIA_TYPE_VIDEO
- && (options->dtmf == AST_SDP_DTMF_RFC_4733 || options->dtmf == AST_SDP_DTMF_AUTO)) {
- i = AST_RTP_DTMF;
- rtp_code = ast_rtp_codecs_payload_code(
- ast_rtp_instance_get_codecs(rtp), 0, NULL, i);
- if (-1 < rtp_code) {
- if (ast_sdp_m_add_format(m_line, options, rtp_code, 0, NULL, i)) {
+ if (ast_sdp_m_add_format(m_line, options, rtp_code, 1, format, 0)) {
ast_sdp_m_free(m_line);
+ ao2_ref(format, -1);
return -1;
}
- snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);
- a_line = ast_sdp_a_alloc("fmtp", tmp);
+ if (ast_format_get_maximum_ms(format)
+ && ((ast_format_get_maximum_ms(format) < max_packet_size)
+ || !max_packet_size)) {
+ max_packet_size = ast_format_get_maximum_ms(format);
+ }
+
+ ao2_ref(format, -1);
+ }
+
+ media_type = ast_stream_get_type(stream);
+ if (media_type != AST_MEDIA_TYPE_VIDEO
+ && (options->dtmf == AST_SDP_DTMF_RFC_4733 || options->dtmf == AST_SDP_DTMF_AUTO)) {
+ i = AST_RTP_DTMF;
+ rtp_code = ast_rtp_codecs_payload_code(
+ ast_rtp_instance_get_codecs(rtp), 0, NULL, i);
+ if (-1 < rtp_code) {
+ if (ast_sdp_m_add_format(m_line, options, rtp_code, 0, NULL, i)) {
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+
+ snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);
+ a_line = ast_sdp_a_alloc("fmtp", tmp);
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ }
+ }
+
+ /* If ptime is set add it as an attribute */
+ min_packet_size = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(rtp));
+ if (!min_packet_size) {
+ min_packet_size = ast_format_cap_get_framing(caps);
+ }
+ if (min_packet_size) {
+ snprintf(tmp, sizeof(tmp), "%d", min_packet_size);
+
+ a_line = ast_sdp_a_alloc("ptime", tmp);
if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
ast_sdp_a_free(a_line);
ast_sdp_m_free(m_line);
return -1;
}
}
- }
- if (ast_sdp_m_get_a_count(m_line) == 0) {
- ast_sdp_m_free(m_line);
- return 0;
- }
+ if (max_packet_size) {
+ snprintf(tmp, sizeof(tmp), "%d", max_packet_size);
+ a_line = ast_sdp_a_alloc("maxptime", tmp);
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ }
- /* If ptime is set add it as an attribute */
- min_packet_size = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(rtp));
- if (!min_packet_size) {
- min_packet_size = ast_format_cap_get_framing(caps);
- }
- if (min_packet_size) {
- snprintf(tmp, sizeof(tmp), "%d", min_packet_size);
-
- a_line = ast_sdp_a_alloc("ptime", tmp);
+ a_line = ast_sdp_a_alloc(ast_sdp_state_get_locally_held(sdp_state, stream_index)
+ ? "sendonly" : "sendrecv", "");
if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
ast_sdp_a_free(a_line);
ast_sdp_m_free(m_line);
return -1;
}
- }
- if (max_packet_size) {
- snprintf(tmp, sizeof(tmp), "%d", max_packet_size);
- a_line = ast_sdp_a_alloc("maxptime", tmp);
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
+ add_ssrc_attributes(m_line, options, rtp);
+ } else {
+ /* Declined/disabled stream */
+ struct ast_sdp_payload *payload;
+ const char *fmt;
+
+ /*
+ * Add a static payload type placeholder to the declined/disabled stream.
+ *
+ * XXX We should use the default payload type in the received offer but
+ * we don't have that available.
+ */
+ switch (ast_stream_get_type(stream)) {
+ default:
+ case AST_MEDIA_TYPE_AUDIO:
+ fmt = "0"; /* ulaw */
+ break;
+ case AST_MEDIA_TYPE_VIDEO:
+ fmt = "31"; /* H.261 */
+ break;
+ }
+ payload = ast_sdp_payload_alloc(fmt);
+ if (!payload || ast_sdp_m_add_payload(m_line, payload)) {
+ ast_sdp_payload_free(payload);
ast_sdp_m_free(m_line);
return -1;
}
}
-
- a_line = ast_sdp_a_alloc(ast_sdp_state_get_locally_held(sdp_state, stream_index) ? "sendonly" : "sendrecv", "");
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
- }
-
- add_ssrc_attributes(m_line, options, rtp);
if (ast_sdp_add_m(sdp, m_line)) {
ast_sdp_m_free(m_line);
@@ -1465,27 +1551,37 @@
struct ast_sdp_m_line *m_line;
struct ast_sdp_payload *payload;
char tmp[64];
- struct ast_sockaddr address_udptl;
struct sdp_state_udptl *udptl;
struct ast_sdp_a_line *a_line;
struct sdp_state_stream *stream_state;
+ int udptl_port;
stream = ast_stream_topology_get_stream(capabilities->topology, stream_index);
- udptl = AST_VECTOR_GET(&capabilities->streams, stream_index)->udptl;
ast_assert(sdp && options && stream);
+ stream_state = AST_VECTOR_GET(&capabilities->streams, stream_index);
+ if (stream_state->udptl) {
+ udptl = stream_state->udptl;
+ } else {
+ /* This is a disabled stream */
+ udptl = NULL;
+ }
+
if (udptl) {
+ struct ast_sockaddr address_udptl;
+
if (ast_sdp_state_get_stream_connection_address(sdp_state, 0, &address_udptl)) {
return -1;
}
+ udptl_port = ast_sockaddr_port(&address_udptl);
} else {
- ast_sockaddr_setnull(&address_udptl);
+ udptl_port = 0;
}
m_line = ast_sdp_m_alloc(
ast_codec_media_type2str(ast_stream_get_type(stream)),
- ast_sockaddr_port(&address_udptl), 1, "udptl", NULL);
+ udptl_port, 1, "udptl", NULL);
if (!m_line) {
return -1;
}
@@ -1497,97 +1593,100 @@
return -1;
}
- stream_state = sdp_state_get_stream(sdp_state, stream_index);
+ if (udptl_port) {
+ /* Stream is not declined/disabled */
+ stream_state = sdp_state_get_stream(sdp_state, stream_index);
- snprintf(tmp, sizeof(tmp), "%u", stream_state->t38_local_params.version);
- a_line = ast_sdp_a_alloc("T38FaxVersion", tmp);
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
- }
-
- snprintf(tmp, sizeof(tmp), "%u", t38_get_rate(stream_state->t38_local_params.rate));
- a_line = ast_sdp_a_alloc("T38FaxMaxBitRate", tmp);
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
- }
-
- if (stream_state->t38_local_params.fill_bit_removal) {
- a_line = ast_sdp_a_alloc("T38FaxFillBitRemoval", "");
+ snprintf(tmp, sizeof(tmp), "%u", stream_state->t38_local_params.version);
+ a_line = ast_sdp_a_alloc("T38FaxVersion", tmp);
if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
ast_sdp_a_free(a_line);
ast_sdp_m_free(m_line);
return -1;
}
- }
- if (stream_state->t38_local_params.transcoding_mmr) {
- a_line = ast_sdp_a_alloc("T38FaxTranscodingMMR", "");
+ snprintf(tmp, sizeof(tmp), "%u", t38_get_rate(stream_state->t38_local_params.rate));
+ a_line = ast_sdp_a_alloc("T38FaxMaxBitRate", tmp);
if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
ast_sdp_a_free(a_line);
ast_sdp_m_free(m_line);
return -1;
}
- }
- if (stream_state->t38_local_params.transcoding_jbig) {
- a_line = ast_sdp_a_alloc("T38FaxTranscodingJBIG", "");
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
+ if (stream_state->t38_local_params.fill_bit_removal) {
+ a_line = ast_sdp_a_alloc("T38FaxFillBitRemoval", "");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
}
- }
- switch (stream_state->t38_local_params.rate_management) {
- case AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF:
- a_line = ast_sdp_a_alloc("T38FaxRateManagement", "transferredTCF");
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
+ if (stream_state->t38_local_params.transcoding_mmr) {
+ a_line = ast_sdp_a_alloc("T38FaxTranscodingMMR", "");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
}
- break;
- case AST_T38_RATE_MANAGEMENT_LOCAL_TCF:
- a_line = ast_sdp_a_alloc("T38FaxRateManagement", "localTCF");
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
- }
- break;
- }
- snprintf(tmp, sizeof(tmp), "%u", ast_udptl_get_local_max_datagram(udptl->instance));
- a_line = ast_sdp_a_alloc("T38FaxMaxDatagram", tmp);
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
- }
+ if (stream_state->t38_local_params.transcoding_jbig) {
+ a_line = ast_sdp_a_alloc("T38FaxTranscodingJBIG", "");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ }
- switch (ast_udptl_get_error_correction_scheme(udptl->instance)) {
- case UDPTL_ERROR_CORRECTION_NONE:
- break;
- case UDPTL_ERROR_CORRECTION_FEC:
- a_line = ast_sdp_a_alloc("T38FaxUdpEC", "t38UDPFEC");
+ switch (stream_state->t38_local_params.rate_management) {
+ case AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF:
+ a_line = ast_sdp_a_alloc("T38FaxRateManagement", "transferredTCF");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ break;
+ case AST_T38_RATE_MANAGEMENT_LOCAL_TCF:
+ a_line = ast_sdp_a_alloc("T38FaxRateManagement", "localTCF");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ break;
+ }
+
+ snprintf(tmp, sizeof(tmp), "%u", ast_udptl_get_local_max_datagram(udptl->instance));
+ a_line = ast_sdp_a_alloc("T38FaxMaxDatagram", tmp);
if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
ast_sdp_a_free(a_line);
ast_sdp_m_free(m_line);
return -1;
}
- break;
- case UDPTL_ERROR_CORRECTION_REDUNDANCY:
- a_line = ast_sdp_a_alloc("T38FaxUdpEC", "t38UDPRedundancy");
- if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
- ast_sdp_a_free(a_line);
- ast_sdp_m_free(m_line);
- return -1;
+
+ switch (ast_udptl_get_error_correction_scheme(udptl->instance)) {
+ case UDPTL_ERROR_CORRECTION_NONE:
+ break;
+ case UDPTL_ERROR_CORRECTION_FEC:
+ a_line = ast_sdp_a_alloc("T38FaxUdpEC", "t38UDPFEC");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ break;
+ case UDPTL_ERROR_CORRECTION_REDUNDANCY:
+ a_line = ast_sdp_a_alloc("T38FaxUdpEC", "t38UDPRedundancy");
+ if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+ ast_sdp_a_free(a_line);
+ ast_sdp_m_free(m_line);
+ return -1;
+ }
+ break;
}
- break;
}
if (ast_sdp_add_m(sdp, m_line)) {
diff --git a/tests/test_sdp.c b/tests/test_sdp.c
index 408888f..79b9e7b 100644
--- a/tests/test_sdp.c
+++ b/tests/test_sdp.c
@@ -630,7 +630,7 @@
}
if (ast_stream_topology_get_count(topology) != 3) {
- ast_test_status_update(test, "Unexpected topology count '%d'. Expecting 2\n",
+ ast_test_status_update(test, "Unexpected topology count '%d'. Expecting 3\n",
ast_stream_topology_get_count(topology));
res = AST_TEST_FAIL;
goto end;
@@ -669,11 +669,9 @@
}
m_line = ast_sdp_get_m(sdp, 0);
-
if (validate_m_line(test, m_line, "audio", 1)) {
return -1;
}
-
if (validate_rtpmap(test, m_line, "PCMU")) {
return -1;
}
@@ -682,26 +680,26 @@
if (!validate_rtpmap(test, m_line, "PCMA")) {
return -1;
}
-
if (!validate_rtpmap(test, m_line, "G722")) {
return -1;
}
-
if (!validate_rtpmap(test, m_line, "opus")) {
return -1;
}
m_line = ast_sdp_get_m(sdp, 1);
-
if (validate_m_line(test, m_line, "video", 1)) {
return -1;
}
-
if (validate_rtpmap(test, m_line, "VP8")) {
return -1;
}
-
if (!validate_rtpmap(test, m_line, "H264")) {
+ return -1;
+ }
+
+ m_line = ast_sdp_get_m(sdp, 2);
+ if (validate_m_line(test, m_line, "image", 1)) {
return -1;
}
@@ -719,10 +717,12 @@
static const struct sdp_format offerer_formats[] = {
{ AST_MEDIA_TYPE_AUDIO, "ulaw,alaw,g722,opus" },
{ AST_MEDIA_TYPE_VIDEO, "h264,vp8" },
+ { AST_MEDIA_TYPE_IMAGE, "t38" },
};
static const struct sdp_format answerer_formats[] = {
{ AST_MEDIA_TYPE_AUDIO, "ulaw" },
{ AST_MEDIA_TYPE_VIDEO, "vp8" },
+ { AST_MEDIA_TYPE_IMAGE, "t38" },
};
switch(cmd) {
@@ -795,8 +795,10 @@
static const struct sdp_format offerer_formats[] = {
{ AST_MEDIA_TYPE_AUDIO, "ulaw,alaw,g722,opus" },
{ AST_MEDIA_TYPE_VIDEO, "h264,vp8" },
+ { AST_MEDIA_TYPE_IMAGE, "t38" },
};
static const struct sdp_format answerer_formats[] = {
+ { AST_MEDIA_TYPE_IMAGE, "t38" },
{ AST_MEDIA_TYPE_VIDEO, "vp8" },
{ AST_MEDIA_TYPE_AUDIO, "ulaw" },
};
@@ -853,6 +855,153 @@
if (validate_merged_sdp(test, answerer_sdp)) {
res = AST_TEST_FAIL;
goto end;
+ }
+
+end:
+ ast_sdp_state_free(sdp_state_offerer);
+ ast_sdp_state_free(sdp_state_answerer);
+
+ return res;
+}
+
+static int validate_merged_sdp_asymmetric(struct ast_test *test, const struct ast_sdp *sdp, int is_offer)
+{
+ struct ast_sdp_m_line *m_line;
+ const char *side = is_offer ? "Offer side" : "Answer side";
+
+ if (!sdp) {
+ ast_test_status_update(test, "%s does not have a SDP\n", side);
+ return -1;
+ }
+
+ /* Stream 0 */
+ m_line = ast_sdp_get_m(sdp, 0);
+ if (validate_m_line(test, m_line, "audio", 1)) {
+ return -1;
+ }
+ if (!m_line->port) {
+ ast_test_status_update(test, "%s stream %d does%s have a port\n", side, 0, "n't");
+ return -1;
+ }
+ if (validate_rtpmap(test, m_line, "PCMU")) {
+ return -1;
+ }
+
+ /* The other audio formats should *NOT* be present */
+ if (!validate_rtpmap(test, m_line, "PCMA")) {
+ return -1;
+ }
+ if (!validate_rtpmap(test, m_line, "G722")) {
+ return -1;
+ }
+ if (!validate_rtpmap(test, m_line, "opus")) {
+ return -1;
+ }
+
+ /* The remaining streams should be declined */
+
+ /* Stream 1 */
+ m_line = ast_sdp_get_m(sdp, 1);
+ if (validate_m_line(test, m_line, "audio", 1)) {
+ return -1;
+ }
+ if (m_line->port) {
+ ast_test_status_update(test, "%s stream %d does%s have a port\n", side, 1, "");
+ return -1;
+ }
+
+ /* Stream 2 */
+ m_line = ast_sdp_get_m(sdp, 2);
+ if (validate_m_line(test, m_line, "video", 1)) {
+ return -1;
+ }
+ if (m_line->port) {
+ ast_test_status_update(test, "%s stream %d does%s have a port\n", side, 2, "");
+ return -1;
+ }
+
+ /* Stream 3 */
+ m_line = ast_sdp_get_m(sdp, 3);
+ if (validate_m_line(test, m_line, "image", 1)) {
+ return -1;
+ }
+ if (m_line->port) {
+ ast_test_status_update(test, "%s stream %d does%s have a port\n", side, 3, "");
+ return -1;
+ }
+
+ return 0;
+}
+
+AST_TEST_DEFINE(sdp_merge_asymmetric)
+{
+ enum ast_test_result_state res = AST_TEST_PASS;
+ struct ast_sdp_state *sdp_state_offerer = NULL;
+ struct ast_sdp_state *sdp_state_answerer = NULL;
+ const struct ast_sdp *offerer_sdp;
+ const struct ast_sdp *answerer_sdp;
+
+ static const struct sdp_format offerer_formats[] = {
+ { AST_MEDIA_TYPE_AUDIO, "ulaw,alaw,g722,opus" },
+ { AST_MEDIA_TYPE_AUDIO, "ulaw" },
+ { AST_MEDIA_TYPE_VIDEO, "h261" },
+ { AST_MEDIA_TYPE_IMAGE, "t38" },
+ };
+ static const struct sdp_format answerer_formats[] = {
+ { AST_MEDIA_TYPE_AUDIO, "ulaw" },
+ };
+
+ switch(cmd) {
+ case TEST_INIT:
+ info->name = "sdp_merge_asymmetric";
+ info->category = "/main/sdp/";
+ info->summary = "Merge two SDPs with an asymmetric number of streams";
+ info->description =
+ "SDP 1 offers a four stream topology: Audio,Audio,Video,T.38\n"
+ "SDP 2 only has a single audio stream topology\n"
+ "We ensure that both local SDPs have the expected stream types and\n"
+ "the expected declined streams";
+ return AST_TEST_NOT_RUN;
+ case TEST_EXECUTE:
+ break;
+ }
+
+ sdp_state_offerer = build_sdp_state(ARRAY_LEN(offerer_formats), offerer_formats, NULL);
+ if (!sdp_state_offerer) {
+ res = AST_TEST_FAIL;
+ goto end;
+ }
+
+ sdp_state_answerer = build_sdp_state(ARRAY_LEN(answerer_formats), answerer_formats, NULL);
+ if (!sdp_state_answerer) {
+ res = AST_TEST_FAIL;
+ goto end;
+ }
+
+ offerer_sdp = ast_sdp_state_get_local_sdp(sdp_state_offerer);
+ if (!offerer_sdp) {
+ res = AST_TEST_FAIL;
+ goto end;
+ }
+
+ ast_sdp_state_set_remote_sdp(sdp_state_answerer, offerer_sdp);
+ answerer_sdp = ast_sdp_state_get_local_sdp(sdp_state_answerer);
+ if (!answerer_sdp) {
+ res = AST_TEST_FAIL;
+ goto end;
+ }
+
+ ast_sdp_state_set_remote_sdp(sdp_state_offerer, answerer_sdp);
+
+#if defined(XXX_TODO_NEED_TO_HANDLE_DECLINED_STREAMS_ON_OFFER_SIDE)
+ /* Get the offerer SDP again because it's now going to be the joint SDP */
+ offerer_sdp = ast_sdp_state_get_local_sdp(sdp_state_offerer);
+ if (validate_merged_sdp_asymmetric(test, offerer_sdp, 1)) {
+ res = AST_TEST_FAIL;
+ }
+#endif
+ if (validate_merged_sdp_asymmetric(test, answerer_sdp, 0)) {
+ res = AST_TEST_FAIL;
}
end:
@@ -976,6 +1125,7 @@
AST_TEST_UNREGISTER(sdp_to_topology);
AST_TEST_UNREGISTER(sdp_merge_symmetric);
AST_TEST_UNREGISTER(sdp_merge_crisscross);
+ AST_TEST_UNREGISTER(sdp_merge_asymmetric);
AST_TEST_UNREGISTER(sdp_ssrc_attributes);
return 0;
@@ -990,6 +1140,7 @@
AST_TEST_REGISTER(sdp_to_topology);
AST_TEST_REGISTER(sdp_merge_symmetric);
AST_TEST_REGISTER(sdp_merge_crisscross);
+ AST_TEST_REGISTER(sdp_merge_asymmetric);
AST_TEST_REGISTER(sdp_ssrc_attributes);
return AST_MODULE_LOAD_SUCCESS;
--
To view, visit https://gerrit.asterisk.org/5598
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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