/*** MODULEINFO
<support_level>core</support_level>
+***/
+
+/*** DOCUMENTATION
+ <managerEvent language="en_US" name="RTCPSent">
+ <managerEventInstance class="EVENT_FLAG_REPORTING">
+ <synopsis>Raised when an RTCP packet is sent.</synopsis>
+ <syntax>
+ <channel_snapshot/>
+ <parameter name="SSRC">
+ <para>The SSRC identifier for our stream</para>
+ </parameter>
+ <parameter name="PT">
+ <para>The type of packet for this RTCP report.</para>
+ <enumlist>
+ <enum name="200(SR)"/>
+ <enum name="201(RR)"/>
+ </enumlist>
+ </parameter>
+ <parameter name="To">
+ <para>The address the report is sent to.</para>
+ </parameter>
+ <parameter name="ReportCount">
+ <para>The number of reports that were sent.</para>
+ <para>The report count determines the number of ReportX headers in
+ the message. The X for each set of report headers will range from 0 to
+ <literal>ReportCount - 1</literal>.</para>
+ </parameter>
+ <parameter name="SentNTP" required="false">
+ <para>The time the sender generated the report. Only valid when
+ PT is <literal>200(SR)</literal>.</para>
+ </parameter>
+ <parameter name="SentRTP" required="false">
+ <para>The sender's last RTP timestamp. Only valid when PT is
+ <literal>200(SR)</literal>.</para>
+ </parameter>
+ <parameter name="SentPackets" required="false">
+ <para>The number of packets the sender has sent. Only valid when PT
+ is <literal>200(SR)</literal>.</para>
+ </parameter>
+ <parameter name="SentOctets" required="false">
+ <para>The number of bytes the sender has sent. Only valid when PT is
+ <literal>200(SR)</literal>.</para>
+ </parameter>
+ <parameter name="ReportXSourceSSRC">
+ <para>The SSRC for the source of this report block.</para>
+ </parameter>
+ <parameter name="ReportXFractionLost">
+ <para>The fraction of RTP data packets from <literal>ReportXSourceSSRC</literal>
+ lost since the previous SR or RR report was sent.</para>
+ </parameter>
+ <parameter name="ReportXCumulativeLost">
+ <para>The total number of RTP data packets from <literal>ReportXSourceSSRC</literal>
+ lost since the beginning of reception.</para>
+ </parameter>
+ <parameter name="ReportXHighestSequence">
+ <para>The highest sequence number received in an RTP data packet from
+ <literal>ReportXSourceSSRC</literal>.</para>
+ </parameter>
+ <parameter name="ReportXSequenceNumberCycles">
+ <para>The number of sequence number cycles seen for the RTP data
+ received from <literal>ReportXSourceSSRC</literal>.</para>
+ </parameter>
+ <parameter name="ReportXIAJitter">
+ <para>An estimate of the statistical variance of the RTP data packet
+ interarrival time, measured in timestamp units.</para>
+ </parameter>
+ <parameter name="ReportXLSR">
+ <para>The last SR timestamp received from <literal>ReportXSourceSSRC</literal>.
+ If no SR has been received from <literal>ReportXSourceSSRC</literal>,
+ then 0.</para>
+ </parameter>
+ <parameter name="ReportXDLSR">
+ <para>The delay, expressed in units of 1/65536 seconds, between
+ receiving the last SR packet from <literal>ReportXSourceSSRC</literal>
+ and sending this report.</para>
+ </parameter>
+ </syntax>
+ <see-also>
+ <ref type="managerEvent">RTCPReceived</ref>
+ </see-also>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="RTCPReceived">
+ <managerEventInstance class="EVENT_FLAG_REPORTING">
+ <synopsis>Raised when an RTCP packet is received.</synopsis>
+ <syntax>
+ <channel_snapshot/>
+ <parameter name="SSRC">
+ <para>The SSRC identifier for the remote system</para>
+ </parameter>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='PT'])" />
+ <parameter name="From">
+ <para>The address the report was received from.</para>
+ </parameter>
+ <parameter name="RTT">
+ <para>Calculated Round-Trip Time in seconds</para>
+ </parameter>
+ <parameter name="ReportCount">
+ <para>The number of reports that were received.</para>
+ <para>The report count determines the number of ReportX headers in
+ the message. The X for each set of report headers will range from 0 to
+ <literal>ReportCount - 1</literal>.</para>
+ </parameter>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentNTP'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentRTP'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentPackets'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentOctets'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[contains(@name, 'ReportX')])" />
+ </syntax>
+ <see-also>
+ <ref type="managerEvent">RTCPSent</ref>
+ </see-also>
+ </managerEventInstance>
+ </managerEvent>
***/
#include "asterisk.h"
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <math.h>
-
-#include "asterisk/channel.h"
-#include "asterisk/frame.h"
-#include "asterisk/module.h"
-#include "asterisk/rtp_engine.h"
+#include <math.h> /* for sqrt, MAX */
+#include <sched.h> /* for sched_yield */
+#include <sys/time.h> /* for timeval */
+#include <time.h> /* for time_t */
+
+#include "asterisk/_private.h" /* for ast_rtp_engine_init prototype */
+#include "asterisk/astobj2.h" /* for ao2_cleanup, ao2_ref, etc */
+#include "asterisk/channel.h" /* for ast_channel_name, etc */
+#include "asterisk/codec.h" /* for ast_codec_media_type2str, etc */
+#include "asterisk/format.h" /* for ast_format_cmp, etc */
+#include "asterisk/format_cache.h" /* for ast_format_adpcm, etc */
+#include "asterisk/format_cap.h" /* for ast_format_cap_alloc, etc */
+#include "asterisk/json.h" /* for ast_json_ref, etc */
+#include "asterisk/linkedlists.h" /* for ast_rtp_engine::<anonymous>, etc */
+#include "asterisk/lock.h" /* for ast_rwlock_unlock, etc */
+#include "asterisk/logger.h" /* for ast_log, ast_debug, etc */
#include "asterisk/manager.h"
-#include "asterisk/options.h"
-#include "asterisk/astobj2.h"
-#include "asterisk/pbx.h"
-#include "asterisk/translate.h"
-#include "asterisk/netsock2.h"
-#include "asterisk/_private.h"
-#include "asterisk/framehook.h"
+#include "asterisk/module.h" /* for ast_module_unref, etc */
+#include "asterisk/netsock2.h" /* for ast_sockaddr_copy, etc */
+#include "asterisk/options.h" /* for ast_option_rtpptdynamic */
+#include "asterisk/pbx.h" /* for pbx_builtin_setvar_helper */
+#include "asterisk/res_srtp.h" /* for ast_srtp_res */
+#include "asterisk/rtp_engine.h" /* for ast_rtp_codecs, etc */
+#include "asterisk/stasis.h" /* for stasis_message_data, etc */
+#include "asterisk/stasis_channels.h" /* for ast_channel_stage_snapshot, etc */
+#include "asterisk/strings.h" /* for ast_str_append, etc */
+#include "asterisk/time.h" /* for ast_tvdiff_ms, ast_tvnow */
+#include "asterisk/translate.h" /* for ast_translate_available_formats */
+#include "asterisk/utils.h" /* for ast_free, ast_strdup, etc */
+#include "asterisk/vector.h" /* for AST_VECTOR_GET, etc */
struct ast_srtp_res *res_srtp = NULL;
struct ast_srtp_policy_res *res_srtp_policy = NULL;
int properties[AST_RTP_PROPERTY_MAX];
/*! Address that we are expecting RTP to come in to */
struct ast_sockaddr local_address;
+ /*! The original source address */
+ struct ast_sockaddr requested_target_address;
/*! Address that we are sending RTP to */
- struct ast_sockaddr remote_address;
- /*! Alternate address that we are receiving RTP from */
- struct ast_sockaddr alt_remote_address;
+ struct ast_sockaddr incoming_source_address;
/*! Instance that we are bridged to if doing remote or local bridging */
struct ast_rtp_instance *bridged;
/*! Payload and packetization information */
int keepalive;
/*! Glue currently in use */
struct ast_rtp_glue *glue;
- /*! Channel associated with the instance */
- struct ast_channel *chan;
/*! SRTP info associated with the instance */
struct ast_srtp *srtp;
+ /*! SRTP info dedicated for RTCP associated with the instance */
+ struct ast_srtp *rtcp_srtp;
+ /*! Channel unique ID */
+ char channel_uniqueid[AST_MAX_UNIQUEID];
+ /*! Time of last packet sent */
+ time_t last_tx;
+ /*! Time of last packet received */
+ time_t last_rx;
};
/*! List of RTP engines that are currently registered */
/*! List of RTP glues */
static AST_RWLIST_HEAD_STATIC(glues, ast_rtp_glue);
+#define MAX_RTP_MIME_TYPES 128
+
/*! The following array defines the MIME Media type (and subtype) for each
of our codecs, or RTP-specific data type. */
static struct ast_rtp_mime_type {
+ /*! \brief A mapping object between the Asterisk codec and this RTP payload */
struct ast_rtp_payload_type payload_type;
- char *type;
- char *subtype;
+ /*! \brief The media type */
+ char type[16];
+ /*! \brief The format type */
+ char subtype[64];
+ /*! \brief Expected sample rate of the /c subtype */
unsigned int sample_rate;
} ast_rtp_mime_types[128]; /* This will Likely not need to grow any time soon. */
static ast_rwlock_t mime_types_lock;
* See http://www.iana.org/assignments/rtp-parameters for a list of
* assigned values
*/
-static struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT];
+static struct ast_rtp_payload_type *static_RTP_PT[AST_RTP_MAX_PT];
static ast_rwlock_t static_RTP_PT_lock;
+/*! \brief \ref stasis topic for RTP related messages */
+static struct stasis_topic *rtp_topic;
+
+
+/*!
+ * \internal
+ * \brief Destructor for \c ast_rtp_payload_type
+ */
+static void rtp_payload_type_dtor(void *obj)
+{
+ struct ast_rtp_payload_type *payload = obj;
+
+ ao2_cleanup(payload->format);
+}
+
+struct ast_rtp_payload_type *ast_rtp_engine_alloc_payload_type(void)
+{
+ struct ast_rtp_payload_type *payload;
+
+ payload = ao2_alloc_options(sizeof(*payload), rtp_payload_type_dtor,
+ AO2_ALLOC_OPT_LOCK_NOLOCK);
+
+ return payload;
+}
+
int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
{
struct ast_rtp_engine *current_engine;
res_srtp->destroy(instance->srtp);
}
+ if (instance->rtcp_srtp) {
+ res_srtp->destroy(instance->rtcp_srtp);
+ }
+
+ ast_rtp_codecs_payloads_destroy(&instance->codecs);
+
/* Drop our engine reference */
ast_module_unref(instance->engine->mod);
ast_sockaddr_copy(&instance->local_address, sa);
ast_sockaddr_copy(&address, sa);
+ if (ast_rtp_codecs_payloads_initialize(&instance->codecs)) {
+ ao2_ref(instance, -1);
+ return NULL;
+ }
+
ast_debug(1, "Using engine '%s' for RTP instance '%p'\n", engine->name, instance);
/* And pass it off to the engine to setup */
return instance;
}
+const char *ast_rtp_instance_get_channel_id(struct ast_rtp_instance *instance)
+{
+ return instance->channel_uniqueid;
+}
+
+void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid)
+{
+ ast_copy_string(instance->channel_uniqueid, uniqueid, sizeof(instance->channel_uniqueid));
+}
+
void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data)
{
instance->data = data;
return 0;
}
-int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance,
- const struct ast_sockaddr *address)
+int ast_rtp_instance_set_incoming_source_address(struct ast_rtp_instance *instance,
+ const struct ast_sockaddr *address)
{
- ast_sockaddr_copy(&instance->remote_address, address);
+ ast_sockaddr_copy(&instance->incoming_source_address, address);
/* moo */
if (instance->engine->remote_address_set) {
- instance->engine->remote_address_set(instance, &instance->remote_address);
+ instance->engine->remote_address_set(instance, &instance->incoming_source_address);
}
return 0;
}
-int ast_rtp_instance_set_alt_remote_address(struct ast_rtp_instance *instance,
- const struct ast_sockaddr *address)
+int ast_rtp_instance_set_requested_target_address(struct ast_rtp_instance *instance,
+ const struct ast_sockaddr *address)
{
- ast_sockaddr_copy(&instance->alt_remote_address, address);
-
- /* oink */
+ ast_sockaddr_copy(&instance->requested_target_address, address);
- if (instance->engine->alt_remote_address_set) {
- instance->engine->alt_remote_address_set(instance, &instance->alt_remote_address);
- }
-
- return 0;
+ return ast_rtp_instance_set_incoming_source_address(instance, address);
}
int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance,
ast_sockaddr_copy(address, &instance->local_address);
}
-int ast_rtp_instance_get_and_cmp_remote_address(struct ast_rtp_instance *instance,
+int ast_rtp_instance_get_and_cmp_requested_target_address(struct ast_rtp_instance *instance,
struct ast_sockaddr *address)
{
- if (ast_sockaddr_cmp(address, &instance->remote_address) != 0) {
- ast_sockaddr_copy(address, &instance->remote_address);
+ if (ast_sockaddr_cmp(address, &instance->requested_target_address) != 0) {
+ ast_sockaddr_copy(address, &instance->requested_target_address);
return 1;
}
return 0;
}
-void ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance,
- struct ast_sockaddr *address)
+void ast_rtp_instance_get_incoming_source_address(struct ast_rtp_instance *instance,
+ struct ast_sockaddr *address)
{
- ast_sockaddr_copy(address, &instance->remote_address);
+ ast_sockaddr_copy(address, &instance->incoming_source_address);
+}
+
+void ast_rtp_instance_get_requested_target_address(struct ast_rtp_instance *instance,
+ struct ast_sockaddr *address)
+{
+ ast_sockaddr_copy(address, &instance->requested_target_address);
}
void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value)
return &instance->codecs;
}
+int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs)
+{
+ int res;
+
+ codecs->framing = 0;
+ ast_rwlock_init(&codecs->codecs_lock);
+ res = AST_VECTOR_INIT(&codecs->payload_mapping_rx, AST_RTP_MAX_PT);
+ res |= AST_VECTOR_INIT(&codecs->payload_mapping_tx, AST_RTP_MAX_PT);
+ if (res) {
+ AST_VECTOR_FREE(&codecs->payload_mapping_rx);
+ AST_VECTOR_FREE(&codecs->payload_mapping_tx);
+ }
+
+ return res;
+}
+
+void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
+{
+ int idx;
+ struct ast_rtp_payload_type *type;
+
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+ ao2_t_cleanup(type, "destroying ast_rtp_codec rx mapping");
+ }
+ AST_VECTOR_FREE(&codecs->payload_mapping_rx);
+
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+ ao2_t_cleanup(type, "destroying ast_rtp_codec tx mapping");
+ }
+ AST_VECTOR_FREE(&codecs->payload_mapping_tx);
+
+ ast_rwlock_destroy(&codecs->codecs_lock);
+}
+
void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
{
- int i;
+ ast_rtp_codecs_payloads_destroy(codecs);
+ ast_rtp_codecs_payloads_initialize(codecs);
- for (i = 0; i < AST_RTP_MAX_PT; i++) {
- codecs->payloads[i].asterisk_format = 0;
- codecs->payloads[i].rtp_code = 0;
- ast_format_clear(&codecs->payloads[i].format);
- if (instance && instance->engine && instance->engine->payload_set) {
+ if (instance && instance->engine && instance->engine->payload_set) {
+ int i;
+ for (i = 0; i < AST_RTP_MAX_PT; i++) {
instance->engine->payload_set(instance, i, 0, NULL, 0);
}
}
}
-void ast_rtp_codecs_payloads_default(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
+/*!
+ * \internal
+ * \brief Clear the rx primary mapping flag on all other matching mappings.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs that need rx clearing.
+ * \param to_match Payload type object to compare against.
+ *
+ * \note It is assumed that codecs is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void payload_mapping_rx_clear_primary(struct ast_rtp_codecs *codecs, struct ast_rtp_payload_type *to_match)
{
- int i;
+ int idx;
+ struct ast_rtp_payload_type *current;
+ struct ast_rtp_payload_type *new_type;
+ struct timeval now;
- ast_rwlock_rdlock(&static_RTP_PT_lock);
- for (i = 0; i < AST_RTP_MAX_PT; i++) {
- if (static_RTP_PT[i].rtp_code || static_RTP_PT[i].asterisk_format) {
+ if (!to_match->primary_mapping) {
+ return;
+ }
- codecs->payloads[i].asterisk_format = static_RTP_PT[i].asterisk_format;
- codecs->payloads[i].rtp_code = static_RTP_PT[i].rtp_code;
- ast_format_copy(&codecs->payloads[i].format, &static_RTP_PT[i].format);
- if (instance && instance->engine && instance->engine->payload_set) {
- instance->engine->payload_set(instance, i, codecs->payloads[i].asterisk_format, &codecs->payloads[i].format, codecs->payloads[i].rtp_code);
+ now = ast_tvnow();
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+ current = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+
+ if (!current || current == to_match || !current->primary_mapping) {
+ continue;
+ }
+ if (current->asterisk_format && to_match->asterisk_format) {
+ if (ast_format_cmp(current->format, to_match->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ continue;
}
+ } else if (!current->asterisk_format && !to_match->asterisk_format) {
+ if (current->rtp_code != to_match->rtp_code) {
+ continue;
+ }
+ } else {
+ continue;
+ }
+
+ /* Replace current with non-primary marked version */
+ new_type = ast_rtp_engine_alloc_payload_type();
+ if (!new_type) {
+ continue;
}
+ *new_type = *current;
+ new_type->primary_mapping = 0;
+ new_type->when_retired = now;
+ ao2_bump(new_type->format);
+ AST_VECTOR_REPLACE(&codecs->payload_mapping_rx, idx, new_type);
+ ao2_ref(current, -1);
}
- ast_rwlock_unlock(&static_RTP_PT_lock);
}
-void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+/*!
+ * \internal
+ * \brief Put the new_type into the rx payload type mapping.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to put new_type into
+ * \param payload type position to replace.
+ * \param new_type RTP payload mapping object to store.
+ *
+ * \note It is assumed that codecs is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void rtp_codecs_payload_replace_rx(struct ast_rtp_codecs *codecs, int payload, struct ast_rtp_payload_type *new_type)
{
- int i;
+ ao2_ref(new_type, +1);
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_rx)) {
+ ao2_t_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_rx, payload),
+ "cleaning up rx mapping vector element about to be replaced");
+ }
+ AST_VECTOR_REPLACE(&codecs->payload_mapping_rx, payload, new_type);
- for (i = 0; i < AST_RTP_MAX_PT; i++) {
- if (src->payloads[i].rtp_code || src->payloads[i].asterisk_format) {
- ast_debug(2, "Copying payload %d from %p to %p\n", i, src, dest);
- dest->payloads[i].asterisk_format = src->payloads[i].asterisk_format;
- dest->payloads[i].rtp_code = src->payloads[i].rtp_code;
- ast_format_copy(&dest->payloads[i].format, &src->payloads[i].format);
- if (instance && instance->engine && instance->engine->payload_set) {
- instance->engine->payload_set(instance, i, dest->payloads[i].asterisk_format, &dest->payloads[i].format, dest->payloads[i].rtp_code);
+ payload_mapping_rx_clear_primary(codecs, new_type);
+}
+
+/*!
+ * \internal
+ * \brief Copy the rx payload type mapping to the destination.
+ * \since 14.0.0
+ *
+ * \param src The source codecs structure
+ * \param dest The destination codecs structure that the values from src will be copied to
+ * \param instance Optionally the instance that the dst codecs structure belongs to
+ *
+ * \note It is assumed that src is at least read locked before calling.
+ * \note It is assumed that dest is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void rtp_codecs_payloads_copy_rx(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+ int idx;
+ struct ast_rtp_payload_type *type;
+
+ for (idx = 0; idx < AST_VECTOR_SIZE(&src->payload_mapping_rx); ++idx) {
+ type = AST_VECTOR_GET(&src->payload_mapping_rx, idx);
+ if (!type) {
+ continue;
+ }
+
+ ast_debug(2, "Copying rx payload mapping %d (%p) from %p to %p\n",
+ idx, type, src, dest);
+ rtp_codecs_payload_replace_rx(dest, idx, type);
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, idx, type->asterisk_format, type->format, type->rtp_code);
+ }
+ }
+}
+
+/*!
+ * \internal
+ * \brief Determine if a type of payload is already present in mappings.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs to be checked for mappings.
+ * \param to_match Payload type object to compare against.
+ *
+ * \note It is assumed that codecs is write locked before calling.
+ *
+ * \retval 0 not found
+ * \retval 1 found
+ */
+static int payload_mapping_tx_is_present(const struct ast_rtp_codecs *codecs, const struct ast_rtp_payload_type *to_match)
+{
+ int idx;
+ struct ast_rtp_payload_type *current;
+
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+ current = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+
+ if (!current) {
+ continue;
+ }
+ if (current == to_match) {
+ /* The exact object is already in the mapping. */
+ return 1;
+ }
+ if (current->asterisk_format && to_match->asterisk_format) {
+ if (ast_format_get_codec_id(current->format) != ast_format_get_codec_id(to_match->format)) {
+ continue;
+ } else if (current->payload == to_match->payload) {
+ return 0;
+ }
+ } else if (!current->asterisk_format && !to_match->asterisk_format) {
+ if (current->rtp_code != to_match->rtp_code) {
+ continue;
}
+ } else {
+ continue;
+ }
+
+ return 1;
+ }
+
+ return 0;
+}
+
+/*!
+ * \internal
+ * \brief Copy the tx payload type mapping to the destination.
+ * \since 14.0.0
+ *
+ * \param src The source codecs structure
+ * \param dest The destination codecs structure that the values from src will be copied to
+ * \param instance Optionally the instance that the dst codecs structure belongs to
+ *
+ * \note It is assumed that src is at least read locked before calling.
+ * \note It is assumed that dest is write locked before calling.
+ *
+ * \return Nothing
+ */
+static void rtp_codecs_payloads_copy_tx(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+ int idx;
+ struct ast_rtp_payload_type *type;
+
+ for (idx = 0; idx < AST_VECTOR_SIZE(&src->payload_mapping_tx); ++idx) {
+ type = AST_VECTOR_GET(&src->payload_mapping_tx, idx);
+ if (!type) {
+ continue;
+ }
+
+ ast_debug(2, "Copying tx payload mapping %d (%p) from %p to %p\n",
+ idx, type, src, dest);
+ ao2_ref(type, +1);
+ if (idx < AST_VECTOR_SIZE(&dest->payload_mapping_tx)) {
+ ao2_t_cleanup(AST_VECTOR_GET(&dest->payload_mapping_tx, idx),
+ "cleaning up tx mapping vector element about to be replaced");
+ }
+ AST_VECTOR_REPLACE(&dest->payload_mapping_tx, idx, type);
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, idx, type->asterisk_format, type->format, type->rtp_code);
+ }
+ }
+}
+
+void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+ int idx;
+ struct ast_rtp_payload_type *type;
+
+ ast_rwlock_wrlock(&dest->codecs_lock);
+
+ /* Deadlock avoidance because of held write lock. */
+ while (ast_rwlock_tryrdlock(&src->codecs_lock)) {
+ ast_rwlock_unlock(&dest->codecs_lock);
+ sched_yield();
+ ast_rwlock_wrlock(&dest->codecs_lock);
+ }
+
+ /*
+ * This represents a completely new mapping of what the remote party is
+ * expecting for payloads, so we clear out the entire tx payload mapping
+ * vector and replace it.
+ */
+ for (idx = 0; idx < AST_VECTOR_SIZE(&dest->payload_mapping_tx); ++idx) {
+ type = AST_VECTOR_GET(&dest->payload_mapping_tx, idx);
+ ao2_t_cleanup(type, "destroying ast_rtp_codec tx mapping");
+ AST_VECTOR_REPLACE(&dest->payload_mapping_tx, idx, NULL);
+ }
+
+ rtp_codecs_payloads_copy_rx(src, dest, instance);
+ rtp_codecs_payloads_copy_tx(src, dest, instance);
+ dest->framing = src->framing;
+
+ ast_rwlock_unlock(&src->codecs_lock);
+ ast_rwlock_unlock(&dest->codecs_lock);
+}
+
+void ast_rtp_codecs_payloads_xover(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
+{
+ int idx;
+ struct ast_rtp_payload_type *type;
+
+ ast_rwlock_wrlock(&dest->codecs_lock);
+ if (src != dest) {
+ /* Deadlock avoidance because of held write lock. */
+ while (ast_rwlock_tryrdlock(&src->codecs_lock)) {
+ ast_rwlock_unlock(&dest->codecs_lock);
+ sched_yield();
+ ast_rwlock_wrlock(&dest->codecs_lock);
+ }
+ }
+
+ /* Crossover copy payload type tx mapping to rx mapping. */
+ for (idx = 0; idx < AST_VECTOR_SIZE(&src->payload_mapping_tx); ++idx) {
+ type = AST_VECTOR_GET(&src->payload_mapping_tx, idx);
+ if (!type) {
+ continue;
}
+
+ /* All tx mapping elements should have the primary flag set. */
+ ast_assert(type->primary_mapping);
+
+ ast_debug(2, "Crossover copying tx to rx payload mapping %d (%p) from %p to %p\n",
+ idx, type, src, dest);
+ rtp_codecs_payload_replace_rx(dest, idx, type);
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, idx, type->asterisk_format, type->format, type->rtp_code);
+ }
+ }
+
+ dest->framing = src->framing;
+
+ if (src != dest) {
+ ast_rwlock_unlock(&src->codecs_lock);
}
+ ast_rwlock_unlock(&dest->codecs_lock);
}
void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
{
+ struct ast_rtp_payload_type *new_type;
+
+ if (payload < 0 || payload >= AST_RTP_MAX_PT) {
+ return;
+ }
ast_rwlock_rdlock(&static_RTP_PT_lock);
- if (payload < 0 || payload >= AST_RTP_MAX_PT || (!static_RTP_PT[payload].rtp_code && !static_RTP_PT[payload].asterisk_format)) {
- ast_rwlock_unlock(&static_RTP_PT_lock);
+ new_type = ao2_bump(static_RTP_PT[payload]);
+ ast_rwlock_unlock(&static_RTP_PT_lock);
+ if (!new_type) {
+ ast_debug(1, "Don't have a default tx payload type %d format for m type on %p\n",
+ payload, codecs);
return;
}
- codecs->payloads[payload].asterisk_format = static_RTP_PT[payload].asterisk_format;
- codecs->payloads[payload].rtp_code = static_RTP_PT[payload].rtp_code;
- ast_format_copy(&codecs->payloads[payload].format, &static_RTP_PT[payload].format);
+ ast_debug(1, "Setting tx payload type %d based on m type on %p\n",
+ payload, codecs);
- ast_debug(1, "Setting payload %d based on m type on %p\n", payload, codecs);
+ ast_rwlock_wrlock(&codecs->codecs_lock);
- if (instance && instance->engine && instance->engine->payload_set) {
- instance->engine->payload_set(instance, payload, codecs->payloads[payload].asterisk_format, &codecs->payloads[payload].format, codecs->payloads[payload].rtp_code);
+ if (!payload_mapping_tx_is_present(codecs, new_type)) {
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+ ao2_t_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_tx, payload),
+ "cleaning up replaced tx payload type");
+ }
+ AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, payload, new_type);
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, payload, new_type->asterisk_format, new_type->format, new_type->rtp_code);
+ }
+ } else {
+ ao2_ref(new_type, -1);
}
- ast_rwlock_unlock(&static_RTP_PT_lock);
+
+ ast_rwlock_unlock(&codecs->codecs_lock);
}
int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
enum ast_rtp_options options,
unsigned int sample_rate)
{
- unsigned int i;
+ unsigned int idx;
int found = 0;
- if (pt < 0 || pt >= AST_RTP_MAX_PT)
+ if (pt < 0 || pt >= AST_RTP_MAX_PT) {
return -1; /* bogus payload type */
+ }
ast_rwlock_rdlock(&mime_types_lock);
- for (i = 0; i < mime_types_len; ++i) {
- const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[i];
+ ast_rwlock_wrlock(&codecs->codecs_lock);
+
+ for (idx = 0; idx < mime_types_len; ++idx) {
+ const struct ast_rtp_mime_type *t = &ast_rtp_mime_types[idx];
+ struct ast_rtp_payload_type *new_type;
if (strcasecmp(mimesubtype, t->subtype)) {
continue;
}
found = 1;
- codecs->payloads[pt] = t->payload_type;
- if ((t->payload_type.format.id == AST_FORMAT_G726) && t->payload_type.asterisk_format && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
- ast_format_set(&codecs->payloads[pt].format, AST_FORMAT_G726_AAL2, 0);
+ new_type = ast_rtp_engine_alloc_payload_type();
+ if (!new_type) {
+ continue;
}
- if (instance && instance->engine && instance->engine->payload_set) {
- instance->engine->payload_set(instance, pt, codecs->payloads[i].asterisk_format, &codecs->payloads[i].format, codecs->payloads[i].rtp_code);
+ new_type->asterisk_format = t->payload_type.asterisk_format;
+ new_type->rtp_code = t->payload_type.rtp_code;
+ new_type->payload = pt;
+ new_type->primary_mapping = 1;
+ if (t->payload_type.asterisk_format
+ && ast_format_cmp(t->payload_type.format, ast_format_g726) == AST_FORMAT_CMP_EQUAL
+ && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+ new_type->format = ast_format_g726_aal2;
+ } else {
+ new_type->format = t->payload_type.format;
+ }
+
+ if (new_type->format) {
+ /* SDP parsing automatically increases the reference count */
+ new_type->format = ast_format_parse_sdp_fmtp(new_type->format, "");
+ }
+
+ if (!payload_mapping_tx_is_present(codecs, new_type)) {
+ if (pt < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+ ao2_t_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_tx, pt),
+ "cleaning up replaced tx payload type");
+ }
+ AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, pt, new_type);
+
+ if (instance && instance->engine && instance->engine->payload_set) {
+ instance->engine->payload_set(instance, pt, new_type->asterisk_format, new_type->format, new_type->rtp_code);
+ }
+ } else {
+ ao2_ref(new_type, -1);
}
break;
}
+
+ ast_rwlock_unlock(&codecs->codecs_lock);
ast_rwlock_unlock(&mime_types_lock);
return (found ? 0 : -2);
void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload)
{
+ struct ast_rtp_payload_type *type;
+
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
return;
}
ast_debug(2, "Unsetting payload %d on %p\n", payload, codecs);
- codecs->payloads[payload].asterisk_format = 0;
- codecs->payloads[payload].rtp_code = 0;
- ast_format_clear(&codecs->payloads[payload].format);
+ ast_rwlock_wrlock(&codecs->codecs_lock);
+
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, payload);
+ ao2_cleanup(type);
+ AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, payload, NULL);
+ }
if (instance && instance->engine && instance->engine->payload_set) {
instance->engine->payload_set(instance, payload, 0, NULL, 0);
}
+
+ ast_rwlock_unlock(&codecs->codecs_lock);
}
-struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload)
+struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *codecs, int payload)
{
- struct ast_rtp_payload_type result = { .asterisk_format = 0, };
+ struct ast_rtp_payload_type *type = NULL;
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
- return result;
+ return NULL;
}
- result.asterisk_format = codecs->payloads[payload].asterisk_format;
- result.rtp_code = codecs->payloads[payload].rtp_code;
- ast_format_copy(&result.format, &codecs->payloads[payload].format);
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_rx)) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_rx, payload);
+ ao2_bump(type);
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
- if (!result.rtp_code && !result.asterisk_format) {
+ if (!type) {
ast_rwlock_rdlock(&static_RTP_PT_lock);
- result = static_RTP_PT[payload];
+ type = ao2_bump(static_RTP_PT[payload]);
ast_rwlock_unlock(&static_RTP_PT_lock);
}
- return result;
+ return type;
}
+int ast_rtp_codecs_payload_replace_format(struct ast_rtp_codecs *codecs, int payload, struct ast_format *format)
+{
+ struct ast_rtp_payload_type *type;
+
+ if (payload < 0 || payload >= AST_RTP_MAX_PT || !format) {
+ return -1;
+ }
+
+ type = ast_rtp_engine_alloc_payload_type();
+ if (!type) {
+ return -1;
+ }
+ ao2_ref(format, +1);
+ type->format = format;
+ type->asterisk_format = 1;
+ type->payload = payload;
+ type->primary_mapping = 1;
+
+ ast_rwlock_wrlock(&codecs->codecs_lock);
+ if (!payload_mapping_tx_is_present(codecs, type)) {
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+ ao2_cleanup(AST_VECTOR_GET(&codecs->payload_mapping_tx, payload));
+ }
+ AST_VECTOR_REPLACE(&codecs->payload_mapping_tx, payload, type);
+ } else {
+ ao2_ref(type, -1);
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+
+ return 0;
+}
struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload)
{
+ struct ast_rtp_payload_type *type;
+ struct ast_format *format = NULL;
+
if (payload < 0 || payload >= AST_RTP_MAX_PT) {
return NULL;
}
- if (!codecs->payloads[payload].asterisk_format) {
- return NULL;
+
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, payload);
+ if (type && type->asterisk_format) {
+ format = ao2_bump(type->format);
+ }
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+
+ return format;
+}
+
+void ast_rtp_codecs_set_framing(struct ast_rtp_codecs *codecs, unsigned int framing)
+{
+ if (!framing) {
+ return;
}
- return &codecs->payloads[payload].format;
+
+ ast_rwlock_wrlock(&codecs->codecs_lock);
+ codecs->framing = framing;
+ ast_rwlock_unlock(&codecs->codecs_lock);
+}
+
+unsigned int ast_rtp_codecs_get_framing(struct ast_rtp_codecs *codecs)
+{
+ unsigned int framing;
+
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ framing = codecs->framing;
+ ast_rwlock_unlock(&codecs->codecs_lock);
+
+ return framing;
}
void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats)
{
- int i;
+ int idx;
- ast_format_cap_remove_all(astformats);
+ ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);
*nonastformats = 0;
- for (i = 0; i < AST_RTP_MAX_PT; i++) {
- if (codecs->payloads[i].rtp_code || codecs->payloads[i].asterisk_format) {
- ast_debug(1, "Incorporating payload %d on %p\n", i, codecs);
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+ struct ast_rtp_payload_type *type;
+
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+ if (!type) {
+ continue;
}
- if (codecs->payloads[i].asterisk_format) {
- ast_format_cap_add(astformats, &codecs->payloads[i].format);
+
+ if (type->asterisk_format) {
+ ast_format_cap_append(astformats, type->format, 0);
} else {
- *nonastformats |= codecs->payloads[i].rtp_code;
+ *nonastformats |= type->rtp_code;
}
}
+ if (codecs->framing) {
+ ast_format_cap_set_framing(astformats, codecs->framing);
+ }
+
+ ast_rwlock_unlock(&codecs->codecs_lock);
}
-int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
+/*!
+ * \internal
+ * \brief Find the static payload type mapping for the format.
+ * \since 14.0.0
+ *
+ * \param asterisk_format Non-zero if the given Asterisk format is present
+ * \param format Asterisk format to look for
+ * \param code The non-Asterisk format code to look for
+ *
+ * \note It is assumed that static_RTP_PT_lock is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if not found.
+ */
+static int find_static_payload_type(int asterisk_format, const struct ast_format *format, int code)
{
- int i;
- int res = -1;
- for (i = 0; i < AST_RTP_MAX_PT; i++) {
- if (codecs->payloads[i].asterisk_format && asterisk_format && format &&
- (ast_format_cmp(format, &codecs->payloads[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
- return i;
- } else if (!codecs->payloads[i].asterisk_format && !asterisk_format &&
- (codecs->payloads[i].rtp_code == code)) {
- return i;
+ int idx;
+ int payload = -1;
+
+ if (!asterisk_format) {
+ for (idx = 0; idx < AST_RTP_MAX_PT; ++idx) {
+ if (static_RTP_PT[idx]
+ && !static_RTP_PT[idx]->asterisk_format
+ && static_RTP_PT[idx]->rtp_code == code) {
+ payload = idx;
+ break;
+ }
+ }
+ } else if (format) {
+ for (idx = 0; idx < AST_RTP_MAX_PT; ++idx) {
+ if (static_RTP_PT[idx]
+ && static_RTP_PT[idx]->asterisk_format
+ && ast_format_cmp(format, static_RTP_PT[idx]->format)
+ != AST_FORMAT_CMP_NOT_EQUAL) {
+ payload = idx;
+ break;
+ }
}
}
- ast_rwlock_rdlock(&static_RTP_PT_lock);
- for (i = 0; i < AST_RTP_MAX_PT; i++) {
- if (static_RTP_PT[i].asterisk_format && asterisk_format && format &&
- (ast_format_cmp(format, &static_RTP_PT[i].format) != AST_FORMAT_CMP_NOT_EQUAL)) {
- res = i;
- break;
- } else if (!static_RTP_PT[i].asterisk_format && !asterisk_format &&
- (static_RTP_PT[i].rtp_code == code)) {
- res = i;
+ return payload;
+}
+
+/*!
+ * \internal
+ * \brief Find the first unused dynamic rx payload type.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to look in
+ *
+ * \note It is assumed that codecs is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if not found.
+ */
+static int rtp_codecs_find_empty_dynamic_rx(struct ast_rtp_codecs *codecs)
+{
+ struct ast_rtp_payload_type *type;
+ int idx;
+ int payload = -1;
+
+ idx = AST_RTP_PT_FIRST_DYNAMIC;
+ for (; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+ if (!type) {
+ payload = idx;
break;
}
}
+ return payload;
+}
+
+/*!
+ * \internal
+ * \brief Find the oldest non-primary dynamic rx payload type.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to look in
+ *
+ * \note It is assumed that codecs is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if not found.
+ */
+static int rtp_codecs_find_non_primary_dynamic_rx(struct ast_rtp_codecs *codecs)
+{
+ struct ast_rtp_payload_type *type;
+ struct timeval oldest;
+ int idx;
+ int payload = -1;
+
+ idx = AST_RTP_PT_FIRST_DYNAMIC;
+ for (; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+ if (type
+ && !type->primary_mapping
+ && (payload == -1
+ || ast_tvdiff_ms(type->when_retired, oldest) < 0)) {
+ oldest = type->when_retired;
+ payload = idx;
+ }
+ }
+ return payload;
+}
+
+/*!
+ * \internal
+ * \brief Assign a payload type for the rx mapping.
+ * \since 14.0.0
+ *
+ * \param codecs Codecs structure to look in
+ * \param asterisk_format Non-zero if the given Asterisk format is present
+ * \param format Asterisk format to look for
+ * \param code The format to look for
+ *
+ * \note It is assumed that static_RTP_PT_lock is at least read locked before calling.
+ *
+ * \retval Numerical payload type
+ * \retval -1 if could not assign.
+ */
+static int rtp_codecs_assign_payload_code_rx(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code)
+{
+ int payload;
+ struct ast_rtp_payload_type *new_type;
+
+ payload = find_static_payload_type(asterisk_format, format, code);
+ if (payload < 0) {
+ return payload;
+ }
+
+ new_type = ast_rtp_engine_alloc_payload_type();
+ if (!new_type) {
+ return -1;
+ }
+ new_type->format = ao2_bump(format);
+ new_type->asterisk_format = asterisk_format;
+ new_type->rtp_code = code;
+ new_type->payload = payload;
+ new_type->primary_mapping = 1;
+
+ ast_rwlock_wrlock(&codecs->codecs_lock);
+ if (payload < AST_RTP_PT_FIRST_DYNAMIC
+ || AST_VECTOR_SIZE(&codecs->payload_mapping_rx) <= payload
+ || !AST_VECTOR_GET(&codecs->payload_mapping_rx, payload)) {
+ /*
+ * The payload type is a static assignment
+ * or our default dynamic position is available.
+ */
+ rtp_codecs_payload_replace_rx(codecs, payload, new_type);
+ } else if (-1 < (payload = rtp_codecs_find_empty_dynamic_rx(codecs))
+ || -1 < (payload = rtp_codecs_find_non_primary_dynamic_rx(codecs))) {
+ /*
+ * We found the first available empty dynamic position
+ * or we found a mapping that should no longer be
+ * actively used.
+ */
+ new_type->payload = payload;
+ rtp_codecs_payload_replace_rx(codecs, payload, new_type);
+ } else {
+ /*
+ * There are no empty or non-primary dynamic positions
+ * left. Sadness.
+ *
+ * I don't think this is really possible.
+ */
+ ast_log(LOG_WARNING, "No dynamic RTP payload type values available!\n");
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+
+ ao2_ref(new_type, -1);
+
+ return payload;
+}
+
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code)
+{
+ struct ast_rtp_payload_type *type;
+ int idx;
+ int payload = -1;
+
+ ast_rwlock_rdlock(&static_RTP_PT_lock);
+ if (!asterisk_format) {
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+ if (!type) {
+ continue;
+ }
+
+ if (!type->asterisk_format
+ && type->primary_mapping
+ && type->rtp_code == code) {
+ payload = idx;
+ break;
+ }
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+ } else if (format) {
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_rx, idx);
+ if (!type) {
+ continue;
+ }
+
+ if (type->asterisk_format
+ && type->primary_mapping
+ && ast_format_cmp(format, type->format) == AST_FORMAT_CMP_EQUAL) {
+ payload = idx;
+ break;
+ }
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+ }
+
+ if (payload < 0) {
+ payload = rtp_codecs_assign_payload_code_rx(codecs, asterisk_format, format,
+ code);
+ }
ast_rwlock_unlock(&static_RTP_PT_lock);
+ return payload;
+}
+
+int ast_rtp_codecs_payload_code_tx(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code)
+{
+ struct ast_rtp_payload_type *type;
+ int idx;
+ int payload = -1;
+
+ if (!asterisk_format) {
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+ if (!type) {
+ continue;
+ }
+
+ if (!type->asterisk_format
+ && type->rtp_code == code) {
+ payload = idx;
+ break;
+ }
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+ } else if (format) {
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ for (idx = 0; idx < AST_VECTOR_SIZE(&codecs->payload_mapping_tx); ++idx) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, idx);
+ if (!type) {
+ continue;
+ }
+
+ if (type->asterisk_format
+ && ast_format_cmp(format, type->format) == AST_FORMAT_CMP_EQUAL) {
+ payload = idx;
+ break;
+ }
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+ }
+
+ if (payload < 0) {
+ ast_rwlock_rdlock(&static_RTP_PT_lock);
+ payload = find_static_payload_type(asterisk_format, format, code);
+ ast_rwlock_unlock(&static_RTP_PT_lock);
+ }
+
+ return payload;
+}
+
+int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int payload)
+{
+ struct ast_rtp_payload_type *type;
+ int res = -1;
+
+ ast_rwlock_rdlock(&codecs->codecs_lock);
+ if (payload < AST_VECTOR_SIZE(&codecs->payload_mapping_tx)) {
+ type = AST_VECTOR_GET(&codecs->payload_mapping_tx, payload);
+ if (type) {
+ res = payload;
+ }
+ }
+ ast_rwlock_unlock(&codecs->codecs_lock);
+
return res;
}
-const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, struct ast_format *format, int code, enum ast_rtp_options options)
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format,
+ const struct ast_format *format, int code, enum ast_rtp_options options)
{
int i;
const char *res = "";
ast_rwlock_rdlock(&mime_types_lock);
for (i = 0; i < mime_types_len; i++) {
if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
- (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
- if ((format->id == AST_FORMAT_G726_AAL2) && (options & AST_RTP_OPT_G726_NONSTANDARD)) {
+ (ast_format_cmp(format, ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
+ if ((ast_format_cmp(format, ast_format_g726_aal2) == AST_FORMAT_CMP_EQUAL) &&
+ (options & AST_RTP_OPT_G726_NONSTANDARD)) {
res = "G726-32";
break;
} else {
return res;
}
-unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format *format, int code)
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format,
+ const struct ast_format *format, int code)
{
unsigned int i;
unsigned int res = 0;
ast_rwlock_rdlock(&mime_types_lock);
for (i = 0; i < mime_types_len; ++i) {
if (ast_rtp_mime_types[i].payload_type.asterisk_format && asterisk_format && format &&
- (ast_format_cmp(format, &ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
+ (ast_format_cmp(format, ast_rtp_mime_types[i].payload_type.format) != AST_FORMAT_CMP_NOT_EQUAL)) {
res = ast_rtp_mime_types[i].sample_rate;
break;
} else if (!ast_rtp_mime_types[i].payload_type.asterisk_format && !asterisk_format &&
if (asterisk_format) {
- struct ast_format tmp_fmt;
- ast_format_cap_iter_start(ast_format_capability);
- while (!ast_format_cap_iter_next(ast_format_capability, &tmp_fmt)) {
- name = ast_rtp_lookup_mime_subtype2(asterisk_format, &tmp_fmt, 0, options);
+ int x;
+ struct ast_format *tmp_fmt;
+ for (x = 0; x < ast_format_cap_count(ast_format_capability); x++) {
+ tmp_fmt = ast_format_cap_get_format(ast_format_capability, x);
+ name = ast_rtp_lookup_mime_subtype2(asterisk_format, tmp_fmt, 0, options);
+ ao2_ref(tmp_fmt, -1);
ast_str_append(&buf, 0, "%s|", name);
found = 1;
}
- ast_format_cap_iter_end(ast_format_capability);
-
} else {
int x;
ast_str_append(&buf, 0, "0x%x (", (unsigned int) rtp_capability);
- for (x = 1; x < AST_RTP_MAX; x <<= 1) {
+ for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
if (rtp_capability & x) {
name = ast_rtp_lookup_mime_subtype2(asterisk_format, NULL, x, options);
ast_str_append(&buf, 0, "%s|", name);
return ast_str_buffer(buf);
}
-void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs)
-{
- codecs->pref = *prefs;
-
- if (instance && instance->engine->packetization_set) {
- instance->engine->packetization_set(instance, &instance->codecs.pref);
- }
-}
-
int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
{
return instance->engine->dtmf_begin ? instance->engine->dtmf_begin(instance, digit) : -1;
return glue;
}
-static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
-{
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
- struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
- struct ast_frame *fr = NULL;
-
- /* Start locally bridging both instances */
- if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
- ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c0), ast_channel_name(c1));
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
- if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
- ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c1), ast_channel_name(c0));
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, NULL);
- }
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
-
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
-
- instance0->bridged = instance1;
- instance1->bridged = instance0;
-
- ast_poll_channel_add(c0, c1);
-
- /* Hop into a loop waiting for a frame from either channel */
- cs[0] = c0;
- cs[1] = c1;
- cs[2] = NULL;
- for (;;) {
- /* If the underlying formats have changed force this bridge to break */
- if ((ast_format_cmp(ast_channel_rawreadformat(c0), ast_channel_rawwriteformat(c1)) == AST_FORMAT_CMP_NOT_EQUAL) ||
- (ast_format_cmp(ast_channel_rawreadformat(c1), ast_channel_rawwriteformat(c0)) == AST_FORMAT_CMP_NOT_EQUAL)) {
- ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
- res = AST_BRIDGE_FAILED_NOWARN;
- break;
- }
- /* Check if anything changed */
- if ((ast_channel_tech_pvt(c0) != pvt0) ||
- (ast_channel_tech_pvt(c1) != pvt1) ||
- (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
- (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
- (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
- ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
- /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
- if ((ast_channel_masq(c0) || ast_channel_masqr(c0)) && (fr = ast_read(c0))) {
- ast_frfree(fr);
- }
- if ((ast_channel_masq(c1) || ast_channel_masqr(c1)) && (fr = ast_read(c1))) {
- ast_frfree(fr);
- }
- res = AST_BRIDGE_RETRY;
- break;
- }
- /* Wait on a channel to feed us a frame */
- if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
- if (!timeoutms) {
- res = AST_BRIDGE_RETRY;
- break;
- }
- ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
- if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
- break;
- }
- continue;
- }
- /* Read in frame from channel */
- fr = ast_read(who);
- other = (who == c0) ? c1 : c0;
- /* Depending on the frame we may need to break out of our bridge */
- if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
- ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
- ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
- /* Record received frame and who */
- *fo = fr;
- *rc = who;
- ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
- res = AST_BRIDGE_COMPLETE;
- break;
- } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
- if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
- (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
- (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
- (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
- (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
- (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
- /* If we are going on hold, then break callback mode and P2P bridging */
- if (fr->subclass.integer == AST_CONTROL_HOLD) {
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, NULL);
- }
- if (instance1->engine->local_bridge) {
- instance1->engine->local_bridge(instance1, NULL);
- }
- instance0->bridged = NULL;
- instance1->bridged = NULL;
- } else if (fr->subclass.integer == AST_CONTROL_UNHOLD) {
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, instance1);
- }
- if (instance1->engine->local_bridge) {
- instance1->engine->local_bridge(instance1, instance0);
- }
- instance0->bridged = instance1;
- instance1->bridged = instance0;
- }
- /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
- if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
- if (ast_channel_connected_line_sub(who, other, fr, 1) &&
- ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
- if (ast_channel_redirecting_sub(who, other, fr, 1) &&
- ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
- ast_channel_hangupcause_hash_set(other, fr->data.ptr);
- ast_frfree(fr);
- } else {
- *fo = fr;
- *rc = who;
- ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
- res = AST_BRIDGE_COMPLETE;
- break;
- }
- } else {
- if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
- (fr->frametype == AST_FRAME_DTMF_END) ||
- (fr->frametype == AST_FRAME_VOICE) ||
- (fr->frametype == AST_FRAME_VIDEO) ||
- (fr->frametype == AST_FRAME_IMAGE) ||
- (fr->frametype == AST_FRAME_HTML) ||
- (fr->frametype == AST_FRAME_MODEM) ||
- (fr->frametype == AST_FRAME_TEXT)) {
- ast_write(other, fr);
- }
-
- ast_frfree(fr);
- }
- /* Swap priority */
- cs[2] = cs[0];
- cs[0] = cs[1];
- cs[1] = cs[2];
- }
-
- /* Stop locally bridging both instances */
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, NULL);
- }
- if (instance1->engine->local_bridge) {
- instance1->engine->local_bridge(instance1, NULL);
- }
-
- instance0->bridged = NULL;
- instance1->bridged = NULL;
-
- ast_poll_channel_del(c0, c1);
-
- return res;
-}
-
-static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0,
- struct ast_channel *c1,
- struct ast_rtp_instance *instance0,
- struct ast_rtp_instance *instance1,
- struct ast_rtp_instance *vinstance0,
- struct ast_rtp_instance *vinstance1,
- struct ast_rtp_instance *tinstance0,
- struct ast_rtp_instance *tinstance1,
- struct ast_rtp_glue *glue0,
- struct ast_rtp_glue *glue1,
- struct ast_format_cap *cap0,
- struct ast_format_cap *cap1,
- int timeoutms,
- int flags,
- struct ast_frame **fo,
- struct ast_channel **rc,
- void *pvt0,
- void *pvt1)
-{
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
- struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
- struct ast_format_cap *oldcap0 = ast_format_cap_dup(cap0);
- struct ast_format_cap *oldcap1 = ast_format_cap_dup(cap1);
- struct ast_sockaddr ac1 = {{0,}}, vac1 = {{0,}}, tac1 = {{0,}}, ac0 = {{0,}}, vac0 = {{0,}}, tac0 = {{0,}};
- struct ast_sockaddr t1 = {{0,}}, vt1 = {{0,}}, tt1 = {{0,}}, t0 = {{0,}}, vt0 = {{0,}}, tt0 = {{0,}};
- struct ast_frame *fr = NULL;
-
- if (!oldcap0 || !oldcap1) {
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- goto remote_bridge_cleanup;
- }
- /* Test the first channel */
- if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0))) {
- ast_rtp_instance_get_remote_address(instance1, &ac1);
- if (vinstance1) {
- ast_rtp_instance_get_remote_address(vinstance1, &vac1);
- }
- if (tinstance1) {
- ast_rtp_instance_get_remote_address(tinstance1, &tac1);
- }
- } else {
- ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
- }
-
- /* Test the second channel */
- if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0))) {
- ast_rtp_instance_get_remote_address(instance0, &ac0);
- if (vinstance0) {
- ast_rtp_instance_get_remote_address(instance0, &vac0);
- }
- if (tinstance0) {
- ast_rtp_instance_get_remote_address(instance0, &tac0);
- }
- } else {
- ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
- }
-
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
-
- instance0->bridged = instance1;
- instance1->bridged = instance0;
-
- ast_poll_channel_add(c0, c1);
-
- /* Go into a loop handling any stray frames that may come in */
- cs[0] = c0;
- cs[1] = c1;
- cs[2] = NULL;
- for (;;) {
- /* Check if anything changed */
- if ((ast_channel_tech_pvt(c0) != pvt0) ||
- (ast_channel_tech_pvt(c1) != pvt1) ||
- (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
- (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
- (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
- ast_debug(1, "Oooh, something is weird, backing out\n");
- res = AST_BRIDGE_RETRY;
- break;
- }
-
- /* Check if they have changed their address */
- ast_rtp_instance_get_remote_address(instance1, &t1);
- if (vinstance1) {
- ast_rtp_instance_get_remote_address(vinstance1, &vt1);
- }
- if (tinstance1) {
- ast_rtp_instance_get_remote_address(tinstance1, &tt1);
- }
- if (glue1->get_codec) {
- ast_format_cap_remove_all(cap1);
- glue1->get_codec(c1, cap1);
- }
-
- ast_rtp_instance_get_remote_address(instance0, &t0);
- if (vinstance0) {
- ast_rtp_instance_get_remote_address(vinstance0, &vt0);
- }
- if (tinstance0) {
- ast_rtp_instance_get_remote_address(tinstance0, &tt0);
- }
- if (glue0->get_codec) {
- ast_format_cap_remove_all(cap0);
- glue0->get_codec(c0, cap0);
- }
-
- if ((ast_sockaddr_cmp(&t1, &ac1)) ||
- (vinstance1 && ast_sockaddr_cmp(&vt1, &vac1)) ||
- (tinstance1 && ast_sockaddr_cmp(&tt1, &tac1)) ||
- (!ast_format_cap_identical(cap1, oldcap1))) {
- char tmp_buf[512] = { 0, };
- ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
- ast_channel_name(c1), ast_sockaddr_stringify(&t1),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
- ast_debug(1, "Oooh, '%s' changed end vaddress to %s (format %s)\n",
- ast_channel_name(c1), ast_sockaddr_stringify(&vt1),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
- ast_debug(1, "Oooh, '%s' changed end taddress to %s (format %s)\n",
- ast_channel_name(c1), ast_sockaddr_stringify(&tt1),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap1));
- ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
- ast_channel_name(c1), ast_sockaddr_stringify(&ac1),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
- ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
- ast_channel_name(c1), ast_sockaddr_stringify(&vac1),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
- ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
- ast_channel_name(c1), ast_sockaddr_stringify(&tac1),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap1));
- if (glue0->update_peer(c0,
- ast_sockaddr_isnull(&t1) ? NULL : instance1,
- ast_sockaddr_isnull(&vt1) ? NULL : vinstance1,
- ast_sockaddr_isnull(&tt1) ? NULL : tinstance1,
- cap1, 0)) {
- ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
- }
- ast_sockaddr_copy(&ac1, &t1);
- ast_sockaddr_copy(&vac1, &vt1);
- ast_sockaddr_copy(&tac1, &tt1);
- ast_format_cap_copy(oldcap1, cap1);
- }
- if ((ast_sockaddr_cmp(&t0, &ac0)) ||
- (vinstance0 && ast_sockaddr_cmp(&vt0, &vac0)) ||
- (tinstance0 && ast_sockaddr_cmp(&tt0, &tac0)) ||
- (!ast_format_cap_identical(cap0, oldcap0))) {
- char tmp_buf[512] = { 0, };
- ast_debug(1, "Oooh, '%s' changed end address to %s (format %s)\n",
- ast_channel_name(c0), ast_sockaddr_stringify(&t0),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), cap0));
- ast_debug(1, "Oooh, '%s' was %s/(format %s)\n",
- ast_channel_name(c0), ast_sockaddr_stringify(&ac0),
- ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), oldcap0));
- if (glue1->update_peer(c1, t0.len ? instance0 : NULL,
- vt0.len ? vinstance0 : NULL,
- tt0.len ? tinstance0 : NULL,
- cap0, 0)) {
- ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
- }
- ast_sockaddr_copy(&ac0, &t0);
- ast_sockaddr_copy(&vac0, &vt0);
- ast_sockaddr_copy(&tac0, &tt0);
- ast_format_cap_copy(oldcap0, cap0);
- }
-
- /* Wait for frame to come in on the channels */
- if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
- if (!timeoutms) {
- res = AST_BRIDGE_RETRY;
- break;
- }
- ast_debug(1, "Ooh, empty read...\n");
- if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
- break;
- }
- continue;
- }
- fr = ast_read(who);
- other = (who == c0) ? c1 : c0;
- if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
- (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
- ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
- /* Break out of bridge */
- *fo = fr;
- *rc = who;
- ast_debug(1, "Oooh, got a %s\n", fr ? "digit" : "hangup");
- res = AST_BRIDGE_COMPLETE;
- break;
- } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
- if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
- (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
- (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
- (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
- (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
- (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
- if (fr->subclass.integer == AST_CONTROL_HOLD) {
- /* If we someone went on hold we want the other side to reinvite back to us */
- if (who == c0) {
- glue1->update_peer(c1, NULL, NULL, NULL, 0, 0);
- } else {
- glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
- }
- } else if (fr->subclass.integer == AST_CONTROL_UNHOLD ||
- fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER) {
- /* If they went off hold they should go back to being direct, or if we have
- * been told to force a peer update, go ahead and do it. */
- if (who == c0) {
- glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0);
- } else {
- glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0);
- }
- }
- /* Update local address information */
- ast_rtp_instance_get_remote_address(instance0, &t0);
- ast_sockaddr_copy(&ac0, &t0);
- ast_rtp_instance_get_remote_address(instance1, &t1);
- ast_sockaddr_copy(&ac1, &t1);
- /* Update codec information */
- if (glue0->get_codec && ast_channel_tech_pvt(c0)) {
- ast_format_cap_remove_all(cap0);
- ast_format_cap_remove_all(oldcap0);
- glue0->get_codec(c0, cap0);
- ast_format_cap_append(oldcap0, cap0);
-
- }
- if (glue1->get_codec && ast_channel_tech_pvt(c1)) {
- ast_format_cap_remove_all(cap1);
- ast_format_cap_remove_all(oldcap1);
- glue1->get_codec(c1, cap1);
- ast_format_cap_append(oldcap1, cap1);
- }
- /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
- if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
- if (ast_channel_connected_line_sub(who, other, fr, 1) &&
- ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
- if (ast_channel_redirecting_sub(who, other, fr, 1) &&
- ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
- ast_channel_hangupcause_hash_set(other, fr->data.ptr);
- ast_frfree(fr);
- } else {
- *fo = fr;
- *rc = who;
- ast_debug(1, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
- res = AST_BRIDGE_COMPLETE;
- goto remote_bridge_cleanup;
- }
- } else {
- if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
- (fr->frametype == AST_FRAME_DTMF_END) ||
- (fr->frametype == AST_FRAME_VOICE) ||
- (fr->frametype == AST_FRAME_VIDEO) ||
- (fr->frametype == AST_FRAME_IMAGE) ||
- (fr->frametype == AST_FRAME_HTML) ||
- (fr->frametype == AST_FRAME_MODEM) ||
- (fr->frametype == AST_FRAME_TEXT)) {
- ast_write(other, fr);
- }
- ast_frfree(fr);
- }
- /* Swap priority */
- cs[2] = cs[0];
- cs[0] = cs[1];
- cs[1] = cs[2];
- }
-
- if (ast_test_flag(ast_channel_flags(c0), AST_FLAG_ZOMBIE)) {
- ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", ast_channel_name(c0));
- } else if (ast_channel_tech_pvt(c0) != pvt0) {
- ast_debug(1, "Channel c0->'%s' pvt changed, in bridge with c1->'%s'\n", ast_channel_name(c0), ast_channel_name(c1));
- } else if (glue0 != ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) {
- ast_debug(1, "Channel c0->'%s' technology changed, in bridge with c1->'%s'\n", ast_channel_name(c0), ast_channel_name(c1));
- } else if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", ast_channel_name(c0));
- }
- if (ast_test_flag(ast_channel_flags(c1), AST_FLAG_ZOMBIE)) {
- ast_debug(1, "Channel '%s' Zombie cleardown from bridge\n", ast_channel_name(c1));
- } else if (ast_channel_tech_pvt(c1) != pvt1) {
- ast_debug(1, "Channel c1->'%s' pvt changed, in bridge with c0->'%s'\n", ast_channel_name(c1), ast_channel_name(c0));
- } else if (glue1 != ast_rtp_instance_get_glue(ast_channel_tech(c1)->type)) {
- ast_debug(1, "Channel c1->'%s' technology changed, in bridge with c0->'%s'\n", ast_channel_name(c1), ast_channel_name(c0));
- } else if (glue1->update_peer(c1, NULL, NULL, NULL, 0, 0)) {
- ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", ast_channel_name(c1));
- }
-
- instance0->bridged = NULL;
- instance1->bridged = NULL;
-
- ast_poll_channel_del(c0, c1);
-
-remote_bridge_cleanup:
- ast_format_cap_destroy(oldcap0);
- ast_format_cap_destroy(oldcap1);
-
- return res;
-}
-
/*!
* \brief Conditionally unref an rtp instance
*/
}
}
-enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
+struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
{
- struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
- *vinstance0 = NULL, *vinstance1 = NULL,
- *tinstance0 = NULL, *tinstance1 = NULL;
- struct ast_rtp_glue *glue0, *glue1;
- struct ast_sockaddr addr1 = { {0, }, }, addr2 = { {0, }, };
- enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
- enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
- enum ast_rtp_dtmf_mode dmode;
- struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
- struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
- int unlock_chans = 1;
-
- if (!cap0 || !cap1) {
- unlock_chans = 0;
- goto done;
- }
-
- /* Lock both channels so we can look for the glue that binds them together */
- ast_channel_lock(c0);
- while (ast_channel_trylock(c1)) {
- ast_channel_unlock(c0);
- usleep(1);
- ast_channel_lock(c0);
- }
-
- /* Ensure neither channel got hungup during lock avoidance */
- if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
- ast_log(LOG_WARNING, "Got hangup while attempting to bridge '%s' and '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
- goto done;
- }
-
- /* Grab glue that binds each channel to something using the RTP engine */
- if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
- ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
- goto done;
- }
-
- audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
- video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
-
- audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
- video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
-
- /* If the channels are of the same technology, they might have limitations on remote bridging */
- if (ast_channel_tech(c0) == ast_channel_tech(c1)) {
- if (audio_glue0_res == audio_glue1_res && audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
- if (glue0->allow_rtp_remote && !(glue0->allow_rtp_remote(c0, c1))) {
- /* If the allow_rtp_remote indicates that remote isn't allowed, revert to local bridge */
- audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
- }
- }
- if (video_glue0_res == video_glue1_res && video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
- if (glue0->allow_vrtp_remote && !(glue0->allow_vrtp_remote(c0, c1))) {
- /* if the allow_vrtp_remote indicates that remote isn't allowed, revert to local bridge */
- video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
- }
- }
- }
-
- /* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
- if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
- audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
- }
- if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
- audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
- }
-
- /* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
- if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID || audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
- res = AST_BRIDGE_FAILED_NOWARN;
- goto done;
- }
-
-
- /* If address families differ, force a local bridge */
- ast_rtp_instance_get_remote_address(instance0, &addr1);
- ast_rtp_instance_get_remote_address(instance1, &addr2);
-
- if (addr1.ss.ss_family != addr2.ss.ss_family ||
- (ast_sockaddr_is_ipv4_mapped(&addr1) != ast_sockaddr_is_ipv4_mapped(&addr2))) {
- audio_glue0_res = AST_RTP_GLUE_RESULT_LOCAL;
- audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
- }
-
- /* If we need to get DTMF see if we can do it outside of the RTP stream itself */
- dmode = ast_rtp_instance_dtmf_mode_get(instance0);
- if ((flags & AST_BRIDGE_DTMF_CHANNEL_0) && dmode) {
- res = AST_BRIDGE_FAILED_NOWARN;
- goto done;
- }
- dmode = ast_rtp_instance_dtmf_mode_get(instance1);
- if ((flags & AST_BRIDGE_DTMF_CHANNEL_1) && dmode) {
- res = AST_BRIDGE_FAILED_NOWARN;
- goto done;
- }
-
- /* If we have gotten to a local bridge make sure that both sides have the same local bridge callback and that they are DTMF compatible */
- if ((audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) && ((instance0->engine->local_bridge != instance1->engine->local_bridge) || (instance0->engine->dtmf_compatible && !instance0->engine->dtmf_compatible(c0, instance0, c1, instance1)))) {
- res = AST_BRIDGE_FAILED_NOWARN;
- goto done;
- }
-
- /* Make sure that codecs match */
- if (glue0->get_codec){
- glue0->get_codec(c0, cap0);
- }
- if (glue1->get_codec) {
- glue1->get_codec(c1, cap1);
- }
- if (!ast_format_cap_is_empty(cap0) && !ast_format_cap_is_empty(cap1) && !ast_format_cap_has_joint(cap0, cap1)) {
- char tmp0[256] = { 0, };
- char tmp1[256] = { 0, };
- ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n",
- ast_getformatname_multiple(tmp0, sizeof(tmp0), cap0),
- ast_getformatname_multiple(tmp1, sizeof(tmp1), cap1));
- res = AST_BRIDGE_FAILED_NOWARN;
- goto done;
- }
-
- instance0->glue = glue0;
- instance1->glue = glue1;
- instance0->chan = c0;
- instance1->chan = c1;
-
- /* Depending on the end result for bridging either do a local bridge or remote bridge */
- if (audio_glue0_res == AST_RTP_GLUE_RESULT_LOCAL || audio_glue1_res == AST_RTP_GLUE_RESULT_LOCAL) {
- ast_verb(3, "Locally bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
- res = local_bridge_loop(c0, c1, instance0, instance1, timeoutms, flags, fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
- } else {
- ast_verb(3, "Remotely bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
- res = remote_bridge_loop(c0, c1, instance0, instance1, vinstance0, vinstance1,
- tinstance0, tinstance1, glue0, glue1, cap0, cap1, timeoutms, flags,
- fo, rc, ast_channel_tech_pvt(c0), ast_channel_tech_pvt(c1));
- }
-
- instance0->glue = NULL;
- instance1->glue = NULL;
- instance0->chan = NULL;
- instance1->chan = NULL;
-
- unlock_chans = 0;
-
-done:
- if (unlock_chans) {
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- }
- ast_format_cap_destroy(cap1);
- ast_format_cap_destroy(cap0);
-
- unref_instance_cond(&instance0);
- unref_instance_cond(&instance1);
- unref_instance_cond(&vinstance0);
- unref_instance_cond(&vinstance1);
- unref_instance_cond(&tinstance0);
- unref_instance_cond(&tinstance1);
-
- return res;
+ return instance->bridged;
}
-struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance)
+void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged)
{
- return instance->bridged;
+ instance->bridged = bridged;
}
-void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1)
+void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src)
{
- struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL,
- *vinstance0 = NULL, *vinstance1 = NULL,
- *tinstance0 = NULL, *tinstance1 = NULL;
- struct ast_rtp_glue *glue0, *glue1;
- enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
- enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
- struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
- struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
+ struct ast_rtp_instance *instance_dst = NULL, *instance_src = NULL,
+ *vinstance_dst = NULL, *vinstance_src = NULL,
+ *tinstance_dst = NULL, *tinstance_src = NULL;
+ struct ast_rtp_glue *glue_dst, *glue_src;
+ enum ast_rtp_glue_result audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
+ enum ast_rtp_glue_result audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID, video_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
+ struct ast_format_cap *cap_dst = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ struct ast_format_cap *cap_src = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
/* Lock both channels so we can look for the glue that binds them together */
- ast_channel_lock_both(c0, c1);
+ ast_channel_lock_both(c_dst, c_src);
- if (!cap1 || !cap0) {
+ if (!cap_src || !cap_dst) {
goto done;
}
/* Grab glue that binds each channel to something using the RTP engine */
- if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
- ast_debug(1, "Can't find native functions for channel '%s'\n", glue0 ? ast_channel_name(c1) : ast_channel_name(c0));
+ if (!(glue_dst = ast_rtp_instance_get_glue(ast_channel_tech(c_dst)->type)) || !(glue_src = ast_rtp_instance_get_glue(ast_channel_tech(c_src)->type))) {
+ ast_debug(1, "Can't find native functions for channel '%s'\n", glue_dst ? ast_channel_name(c_src) : ast_channel_name(c_dst));
goto done;
}
- audio_glue0_res = glue0->get_rtp_info(c0, &instance0);
- video_glue0_res = glue0->get_vrtp_info ? glue0->get_vrtp_info(c0, &vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
+ audio_glue_dst_res = glue_dst->get_rtp_info(c_dst, &instance_dst);
+ video_glue_dst_res = glue_dst->get_vrtp_info ? glue_dst->get_vrtp_info(c_dst, &vinstance_dst) : AST_RTP_GLUE_RESULT_FORBID;
- audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
- video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
+ audio_glue_src_res = glue_src->get_rtp_info(c_src, &instance_src);
+ video_glue_src_res = glue_src->get_vrtp_info ? glue_src->get_vrtp_info(c_src, &vinstance_src) : AST_RTP_GLUE_RESULT_FORBID;
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
- if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
- audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ if (video_glue_dst_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue_dst_res = AST_RTP_GLUE_RESULT_FORBID;
}
- if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
- audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+ if (video_glue_src_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE)) {
+ audio_glue_src_res = AST_RTP_GLUE_RESULT_FORBID;
}
- if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec) {
- glue0->get_codec(c0, cap0);
+ if (audio_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_dst_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_dst_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_dst->get_codec) {
+ glue_dst->get_codec(c_dst, cap_dst);
}
- if (audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue1_res == AST_RTP_GLUE_RESULT_FORBID || video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) && glue1->get_codec) {
- glue1->get_codec(c1, cap1);
+ if (audio_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue_src_res == AST_RTP_GLUE_RESULT_FORBID || video_glue_src_res == AST_RTP_GLUE_RESULT_REMOTE) && glue_src->get_codec) {
+ glue_src->get_codec(c_src, cap_src);
}
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
- if (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE) {
+ if (audio_glue_dst_res != AST_RTP_GLUE_RESULT_REMOTE || audio_glue_src_res != AST_RTP_GLUE_RESULT_REMOTE) {
goto done;
}
/* Make sure we have matching codecs */
- if (!ast_format_cap_has_joint(cap0, cap1)) {
+ if (!ast_format_cap_iscompatible(cap_dst, cap_src)) {
goto done;
}
- ast_rtp_codecs_payloads_copy(&instance0->codecs, &instance1->codecs, instance1);
+ ast_rtp_codecs_payloads_xover(&instance_src->codecs, &instance_dst->codecs, instance_dst);
- if (vinstance0 && vinstance1) {
- ast_rtp_codecs_payloads_copy(&vinstance0->codecs, &vinstance1->codecs, vinstance1);
+ if (vinstance_dst && vinstance_src) {
+ ast_rtp_codecs_payloads_xover(&vinstance_src->codecs, &vinstance_dst->codecs, vinstance_dst);
}
- if (tinstance0 && tinstance1) {
- ast_rtp_codecs_payloads_copy(&tinstance0->codecs, &tinstance1->codecs, tinstance1);
+ if (tinstance_dst && tinstance_src) {
+ ast_rtp_codecs_payloads_xover(&tinstance_src->codecs, &tinstance_dst->codecs, tinstance_dst);
}
- if (glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0)) {
+ if (glue_dst->update_peer(c_dst, instance_src, vinstance_src, tinstance_src, cap_src, 0)) {
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n",
- ast_channel_name(c0), ast_channel_name(c1));
+ ast_channel_name(c_dst), ast_channel_name(c_src));
} else {
ast_debug(1, "Seeded SDP of '%s' with that of '%s'\n",
- ast_channel_name(c0), ast_channel_name(c1));
+ ast_channel_name(c_dst), ast_channel_name(c_src));
}
done:
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
+ ast_channel_unlock(c_dst);
+ ast_channel_unlock(c_src);
- ast_format_cap_destroy(cap0);
- ast_format_cap_destroy(cap1);
+ ao2_cleanup(cap_dst);
+ ao2_cleanup(cap_src);
- unref_instance_cond(&instance0);
- unref_instance_cond(&instance1);
- unref_instance_cond(&vinstance0);
- unref_instance_cond(&vinstance1);
- unref_instance_cond(&tinstance0);
- unref_instance_cond(&tinstance1);
+ unref_instance_cond(&instance_dst);
+ unref_instance_cond(&instance_src);
+ unref_instance_cond(&vinstance_dst);
+ unref_instance_cond(&vinstance_src);
+ unref_instance_cond(&tinstance_dst);
+ unref_instance_cond(&tinstance_src);
}
int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
struct ast_rtp_glue *glue0, *glue1;
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
- struct ast_format_cap *cap0 = ast_format_cap_alloc_nolock();
- struct ast_format_cap *cap1 = ast_format_cap_alloc_nolock();
- int res = 0;
+ struct ast_format_cap *cap0 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ struct ast_format_cap *cap1 = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
/* If there is no second channel just immediately bail out, we are of no use in that scenario */
- if (!c1) {
- ast_format_cap_destroy(cap0);
- ast_format_cap_destroy(cap1);
+ if (!c1 || !cap1 || !cap0) {
+ ao2_cleanup(cap0);
+ ao2_cleanup(cap1);
return -1;
}
/* Lock both channels so we can look for the glue that binds them together */
- ast_channel_lock(c0);
- while (ast_channel_trylock(c1)) {
- ast_channel_unlock(c0);
- usleep(1);
- ast_channel_lock(c0);
- }
-
- if (!cap1 || !cap0) {
- goto done;
- }
+ ast_channel_lock_both(c0, c1);
/* Grab glue that binds each channel to something using the RTP engine */
if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) || !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
}
/* Make sure we have matching codecs */
- if (!ast_format_cap_has_joint(cap0, cap1)) {
+ if (!ast_format_cap_iscompatible(cap0, cap1)) {
goto done;
}
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
}
- res = 0;
-
done:
ast_channel_unlock(c0);
ast_channel_unlock(c1);
- ast_format_cap_destroy(cap0);
- ast_format_cap_destroy(cap1);
+ ao2_cleanup(cap0);
+ ao2_cleanup(cap1);
unref_instance_cond(&instance0);
unref_instance_cond(&instance1);
unref_instance_cond(&tinstance0);
unref_instance_cond(&tinstance1);
- if (!res) {
- ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
- }
+ ast_debug(1, "Setting early bridge SDP of '%s' with that of '%s'\n", ast_channel_name(c0), c1 ? ast_channel_name(c1) : "<unspecified>");
- return res;
+ return 0;
}
int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
/* Now actually fill the buffer with the good information */
if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY) {
- snprintf(buf, size, "ssrc=%i;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
- stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.txjitter, stats.rxcount, stats.rxjitter, stats.txcount, stats.txploss, stats.rtt);
+ snprintf(buf, size, "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f",
+ stats.local_ssrc, stats.remote_ssrc, stats.rxploss, stats.rxjitter, stats.rxcount, stats.txjitter, stats.txcount, stats.txploss, stats.rtt);
} else if (field == AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER) {
snprintf(buf, size, "minrxjitter=%f;maxrxjitter=%f;avgrxjitter=%f;stdevrxjitter=%f;reported_minjitter=%f;reported_maxjitter=%f;reported_avgjitter=%f;reported_stdevjitter=%f;",
stats.local_minjitter, stats.local_maxjitter, stats.local_normdevjitter, sqrt(stats.local_stdevjitter), stats.remote_minjitter, stats.remote_maxjitter, stats.remote_normdevjitter, sqrt(stats.remote_stdevjitter));
void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
{
- char quality_buf[AST_MAX_USER_FIELD], *quality;
- struct ast_channel *bridge = ast_bridged_channel(chan);
+ char quality_buf[AST_MAX_USER_FIELD];
+ char *quality;
+ struct ast_channel *bridge;
+
+ bridge = ast_channel_bridge_peer(chan);
+ if (bridge) {
+ ast_channel_lock_both(chan, bridge);
+ ast_channel_stage_snapshot(bridge);
+ } else {
+ ast_channel_lock(chan);
+ }
+ ast_channel_stage_snapshot(chan);
- if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY,
+ quality_buf, sizeof(quality_buf));
+ if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
}
}
- if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+ quality = ast_rtp_instance_get_quality(instance,
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf));
+ if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
}
}
- if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+ quality = ast_rtp_instance_get_quality(instance,
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf));
+ if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
}
}
- if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+ quality = ast_rtp_instance_get_quality(instance,
+ AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf));
+ if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
}
}
+
+ ast_channel_stage_snapshot_done(chan);
+ ast_channel_unlock(chan);
+ if (bridge) {
+ ast_channel_stage_snapshot_done(bridge);
+ ast_channel_unlock(bridge);
+ ast_channel_unref(bridge);
+ }
}
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)
}
glue->get_rtp_info(peer, &peer_instance);
-
- if (!peer_instance || peer_instance->engine != instance->engine) {
+ if (!peer_instance) {
+ ast_log(LOG_ERROR, "Unable to get_rtp_info for peer type %s\n", glue->type);
+ ast_channel_unlock(peer);
+ return -1;
+ }
+ if (peer_instance->engine != instance->engine) {
+ ast_log(LOG_ERROR, "Peer engine mismatch for type %s\n", glue->type);
ast_channel_unlock(peer);
ao2_ref(peer_instance, -1);
- peer_instance = NULL;
return -1;
}
{
if (instance->engine->available_formats) {
instance->engine->available_formats(instance, to_endpoint, to_asterisk, result);
- if (!ast_format_cap_is_empty(result)) {
+ if (ast_format_cap_count(result)) {
return;
}
}
return instance->glue;
}
-struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance)
-{
- return instance->chan;
-}
-
int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res)
{
if (res_srtp || res_srtp_policy) {
return res_srtp && res_srtp_policy;
}
-int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy)
+int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *remote_policy, struct ast_srtp_policy *local_policy, int rtcp)
{
int res = 0;
+ struct ast_srtp **srtp;
if (!res_srtp) {
return -1;
}
- if (!instance->srtp) {
- res = res_srtp->create(&instance->srtp, instance, remote_policy);
+
+ srtp = rtcp ? &instance->rtcp_srtp : &instance->srtp;
+
+ if (!*srtp) {
+ res = res_srtp->create(srtp, instance, remote_policy);
} else {
- res = res_srtp->replace(&instance->srtp, instance, remote_policy);
+ res = res_srtp->replace(srtp, instance, remote_policy);
}
if (!res) {
- res = res_srtp->add_stream(instance->srtp, local_policy);
+ res = res_srtp->add_stream(*srtp, local_policy);
}
return res;
}
-struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
+struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance, int rtcp)
{
- return instance->srtp;
+ if (rtcp && instance->rtcp_srtp) {
+ return instance->rtcp_srtp;
+ }
+ else {
+ return instance->srtp;
+ }
}
int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
return -1;
}
-static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
+struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance)
{
- int x = mime_types_len;
- if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
- return;
+ return instance->engine->ice;
+}
+
+struct ast_rtp_engine_dtls *ast_rtp_instance_get_dtls(struct ast_rtp_instance *instance)
+{
+ return instance->engine->dtls;
+}
+
+int ast_rtp_dtls_cfg_parse(struct ast_rtp_dtls_cfg *dtls_cfg, const char *name, const char *value)
+{
+ if (!strcasecmp(name, "dtlsenable")) {
+ dtls_cfg->enabled = ast_true(value) ? 1 : 0;
+ } else if (!strcasecmp(name, "dtlsverify")) {
+ if (!strcasecmp(value, "yes")) {
+ dtls_cfg->verify = AST_RTP_DTLS_VERIFY_FINGERPRINT | AST_RTP_DTLS_VERIFY_CERTIFICATE;
+ } else if (!strcasecmp(value, "fingerprint")) {
+ dtls_cfg->verify = AST_RTP_DTLS_VERIFY_FINGERPRINT;
+ } else if (!strcasecmp(value, "certificate")) {
+ dtls_cfg->verify = AST_RTP_DTLS_VERIFY_CERTIFICATE;
+ } else if (!strcasecmp(value, "no")) {
+ dtls_cfg->verify = AST_RTP_DTLS_VERIFY_NONE;
+ } else {
+ return -1;
+ }
+ } else if (!strcasecmp(name, "dtlsrekey")) {
+ if (sscanf(value, "%30u", &dtls_cfg->rekey) != 1) {
+ return -1;
+ }
+ } else if (!strcasecmp(name, "dtlscertfile")) {
+ ast_free(dtls_cfg->certfile);
+ if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+ ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+ return -1;
+ }
+ dtls_cfg->certfile = ast_strdup(value);
+ } else if (!strcasecmp(name, "dtlsprivatekey")) {
+ ast_free(dtls_cfg->pvtfile);
+ if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+ ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+ return -1;
+ }
+ dtls_cfg->pvtfile = ast_strdup(value);
+ } else if (!strcasecmp(name, "dtlscipher")) {
+ ast_free(dtls_cfg->cipher);
+ dtls_cfg->cipher = ast_strdup(value);
+ } else if (!strcasecmp(name, "dtlscafile")) {
+ ast_free(dtls_cfg->cafile);
+ if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+ ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+ return -1;
+ }
+ dtls_cfg->cafile = ast_strdup(value);
+ } else if (!strcasecmp(name, "dtlscapath") || !strcasecmp(name, "dtlscadir")) {
+ ast_free(dtls_cfg->capath);
+ if (!ast_strlen_zero(value) && !ast_file_is_readable(value)) {
+ ast_log(LOG_ERROR, "%s file %s does not exist or is not readable\n", name, value);
+ return -1;
+ }
+ dtls_cfg->capath = ast_strdup(value);
+ } else if (!strcasecmp(name, "dtlssetup")) {
+ if (!strcasecmp(value, "active")) {
+ dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTIVE;
+ } else if (!strcasecmp(value, "passive")) {
+ dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_PASSIVE;
+ } else if (!strcasecmp(value, "actpass")) {
+ dtls_cfg->default_setup = AST_RTP_DTLS_SETUP_ACTPASS;
+ }
+ } else if (!strcasecmp(name, "dtlsfingerprint")) {
+ if (!strcasecmp(value, "sha-256")) {
+ dtls_cfg->hash = AST_RTP_DTLS_HASH_SHA256;
+ } else if (!strcasecmp(value, "sha-1")) {
+ dtls_cfg->hash = AST_RTP_DTLS_HASH_SHA1;
+ }
+ } else {
+ return -1;
}
+ return 0;
+}
+
+void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rtp_dtls_cfg *dst_cfg)
+{
+ ast_rtp_dtls_cfg_free(dst_cfg); /* Prevent a double-call leaking memory via ast_strdup */
+
+ dst_cfg->enabled = src_cfg->enabled;
+ dst_cfg->verify = src_cfg->verify;
+ dst_cfg->rekey = src_cfg->rekey;
+ dst_cfg->suite = src_cfg->suite;
+ dst_cfg->hash = src_cfg->hash;
+ dst_cfg->certfile = ast_strdup(src_cfg->certfile);
+ dst_cfg->pvtfile = ast_strdup(src_cfg->pvtfile);
+ dst_cfg->cipher = ast_strdup(src_cfg->cipher);
+ dst_cfg->cafile = ast_strdup(src_cfg->cafile);
+ dst_cfg->capath = ast_strdup(src_cfg->capath);
+ dst_cfg->default_setup = src_cfg->default_setup;
+}
+
+void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg)
+{
+ ast_free(dtls_cfg->certfile);
+ dtls_cfg->certfile = NULL;
+ ast_free(dtls_cfg->pvtfile);
+ dtls_cfg->pvtfile = NULL;
+ ast_free(dtls_cfg->cipher);
+ dtls_cfg->cipher = NULL;
+ ast_free(dtls_cfg->cafile);
+ dtls_cfg->cafile = NULL;
+ ast_free(dtls_cfg->capath);
+ dtls_cfg->capath = NULL;
+}
+
+/*! \internal
+ * \brief Small helper routine that cleans up entry i in
+ * \c ast_rtp_mime_types.
+ */
+static void rtp_engine_mime_type_cleanup(int i)
+{
+ ao2_cleanup(ast_rtp_mime_types[i].payload_type.format);
+ memset(&ast_rtp_mime_types[i], 0, sizeof(struct ast_rtp_mime_type));
+}
+
+static void set_next_mime_type(struct ast_format *format, int rtp_code, const char *type, const char *subtype, unsigned int sample_rate)
+{
+ int x;
+
ast_rwlock_wrlock(&mime_types_lock);
+
+ x = mime_types_len;
+ if (ARRAY_LEN(ast_rtp_mime_types) <= x) {
+ ast_rwlock_unlock(&mime_types_lock);
+ return;
+ }
+
+ /* Make sure any previous value in ast_rtp_mime_types is cleaned up */
+ memset(&ast_rtp_mime_types[x], 0, sizeof(struct ast_rtp_mime_type));
if (format) {
ast_rtp_mime_types[x].payload_type.asterisk_format = 1;
- ast_format_copy(&ast_rtp_mime_types[x].payload_type.format, format);
+ ast_rtp_mime_types[x].payload_type.format = ao2_bump(format);
} else {
ast_rtp_mime_types[x].payload_type.rtp_code = rtp_code;
}
- ast_rtp_mime_types[x].type = type;
- ast_rtp_mime_types[x].subtype = subtype;
+ ast_copy_string(ast_rtp_mime_types[x].type, type, sizeof(ast_rtp_mime_types[x].type));
+ ast_copy_string(ast_rtp_mime_types[x].subtype, subtype, sizeof(ast_rtp_mime_types[x].subtype));
ast_rtp_mime_types[x].sample_rate = sample_rate;
mime_types_len++;
+
ast_rwlock_unlock(&mime_types_lock);
}
-static void add_static_payload(int map, const struct ast_format *format, int rtp_code)
+static void add_static_payload(int map, struct ast_format *format, int rtp_code)
{
int x;
+ struct ast_rtp_payload_type *type;
+
+ /*
+ * ARRAY_LEN's result is cast to an int so 'map' is not autocast to a size_t,
+ * which if negative would cause an assertion.
+ */
+ ast_assert(map < (int)ARRAY_LEN(static_RTP_PT));
+
ast_rwlock_wrlock(&static_RTP_PT_lock);
if (map < 0) {
/* find next available dynamic payload slot */
- for (x = 96; x < 127; x++) {
- if (!static_RTP_PT[x].asterisk_format && !static_RTP_PT[x].rtp_code) {
+ for (x = AST_RTP_PT_FIRST_DYNAMIC; x < AST_RTP_MAX_PT; ++x) {
+ if (!static_RTP_PT[x]) {
map = x;
break;
}
}
- }
- if (map < 0) {
- ast_log(LOG_WARNING, "No Dynamic RTP mapping avaliable for format %s\n" ,ast_getformatname(format));
- ast_rwlock_unlock(&static_RTP_PT_lock);
- return;
+ /* http://www.iana.org/assignments/rtp-parameters
+ * RFC 3551, Section 3: "[...] applications which need to define more
+ * than 32 dynamic payload types MAY bind codes below 96, in which case
+ * it is RECOMMENDED that unassigned payload type numbers be used
+ * first". Updated by RFC 5761, Section 4: "[...] values in the range
+ * 64-95 MUST NOT be used [to avoid conflicts with RTCP]". Summaries:
+ * https://tools.ietf.org/html/draft-roach-mmusic-unified-plan#section-3.2.1.2
+ * https://tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage#section-3
+ */
+ if (map < 0) {
+ for (x = MAX(ast_option_rtpptdynamic, 35); x <= AST_RTP_PT_LAST_REASSIGN; ++x) {
+ if (!static_RTP_PT[x]) {
+ map = x;
+ break;
+ }
+ }
+ }
+ /* Yet, reusing mappings below 35 is not supported in Asterisk because
+ * when Compact Headers are activated, no rtpmap is send for those below
+ * 35. If you want to use 35 and below
+ * A) do not use Compact Headers,
+ * B) remove that code in chan_sip/res_pjsip, or
+ * C) add a flag that this RTP Payload Type got reassigned dynamically
+ * and requires a rtpmap even with Compact Headers enabled.
+ */
+ if (map < 0) {
+ for (x = MAX(ast_option_rtpptdynamic, 20); x < 35; ++x) {
+ if (!static_RTP_PT[x]) {
+ map = x;
+ break;
+ }
+ }
+ }
+ if (map < 0) {
+ for (x = MAX(ast_option_rtpptdynamic, 0); x < 20; ++x) {
+ if (!static_RTP_PT[x]) {
+ map = x;
+ break;
+ }
+ }
+ }
+
+ if (map < 0) {
+ if (format) {
+ ast_log(LOG_WARNING, "No Dynamic RTP mapping available for format %s\n",
+ ast_format_get_name(format));
+ } else {
+ ast_log(LOG_WARNING, "No Dynamic RTP mapping available for RTP code %d\n",
+ rtp_code);
+ }
+ ast_rwlock_unlock(&static_RTP_PT_lock);
+ return;
+ }
}
- if (format) {
- static_RTP_PT[map].asterisk_format = 1;
- ast_format_copy(&static_RTP_PT[map].format, format);
- } else {
- static_RTP_PT[map].rtp_code = rtp_code;
+ type = ast_rtp_engine_alloc_payload_type();
+ if (type) {
+ if (format) {
+ ao2_ref(format, +1);
+ type->format = format;
+ type->asterisk_format = 1;
+ } else {
+ type->rtp_code = rtp_code;
+ }
+ type->payload = map;
+ type->primary_mapping = 1;
+ ao2_cleanup(static_RTP_PT[map]);
+ static_RTP_PT[map] = type;
}
ast_rwlock_unlock(&static_RTP_PT_lock);
}
-int ast_rtp_engine_load_format(const struct ast_format *format)
+int ast_rtp_engine_load_format(struct ast_format *format)
{
- switch (format->id) {
- case AST_FORMAT_SILK:
- set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
- add_static_payload(-1, format, 0);
- break;
- case AST_FORMAT_CELT:
- set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
- add_static_payload(-1, format, 0);
- break;
- default:
- break;
- }
+ set_next_mime_type(format,
+ 0,
+ ast_codec_media_type2str(ast_format_get_type(format)),
+ ast_format_get_codec_name(format),
+ ast_format_get_sample_rate(format));
+ add_static_payload(-1, format, 0);
return 0;
}
-int ast_rtp_engine_unload_format(const struct ast_format *format)
+int ast_rtp_engine_unload_format(struct ast_format *format)
{
int x;
int y = 0;
ast_rwlock_wrlock(&static_RTP_PT_lock);
/* remove everything pertaining to this format id from the lists */
for (x = 0; x < AST_RTP_MAX_PT; x++) {
- if (ast_format_cmp(&static_RTP_PT[x].format, format) == AST_FORMAT_CMP_EQUAL) {
- memset(&static_RTP_PT[x], 0, sizeof(struct ast_rtp_payload_type));
+ if (static_RTP_PT[x]
+ && ast_format_cmp(static_RTP_PT[x]->format, format) == AST_FORMAT_CMP_EQUAL) {
+ ao2_ref(static_RTP_PT[x], -1);
+ static_RTP_PT[x] = NULL;
}
}
ast_rwlock_unlock(&static_RTP_PT_lock);
-
ast_rwlock_wrlock(&mime_types_lock);
/* rebuild the list skipping the items matching this id */
for (x = 0; x < mime_types_len; x++) {
- if (ast_format_cmp(&ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
+ if (ast_format_cmp(ast_rtp_mime_types[x].payload_type.format, format) == AST_FORMAT_CMP_EQUAL) {
+ rtp_engine_mime_type_cleanup(x);
continue;
}
- ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
+ if (x != y) {
+ ast_rtp_mime_types[y] = ast_rtp_mime_types[x];
+ }
y++;
}
mime_types_len = y;
return 0;
}
-int ast_rtp_engine_init()
+/*!
+ * \internal
+ * \brief \ref stasis message payload for RTCP messages
+ */
+struct rtcp_message_payload {
+ struct ast_channel_snapshot *snapshot; /*< The channel snapshot, if available */
+ struct ast_rtp_rtcp_report *report; /*< The RTCP report */
+ struct ast_json *blob; /*< Extra JSON data to publish */
+};
+
+static void rtcp_message_payload_dtor(void *obj)
+{
+ struct rtcp_message_payload *payload = obj;
+
+ ao2_cleanup(payload->report);
+ ao2_cleanup(payload->snapshot);
+ ast_json_unref(payload->blob);
+}
+
+static struct ast_manager_event_blob *rtcp_report_to_ami(struct stasis_message *msg)
{
- struct ast_format tmpfmt;
+ struct rtcp_message_payload *payload = stasis_message_data(msg);
+ RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
+ RAII_VAR(struct ast_str *, packet_string, ast_str_create(512), ast_free);
+ unsigned int ssrc = payload->report->ssrc;
+ unsigned int type = payload->report->type;
+ unsigned int report_count = payload->report->reception_report_count;
+ int i;
+
+ if (!packet_string) {
+ return NULL;
+ }
+
+ if (payload->snapshot) {
+ channel_string = ast_manager_build_channel_state_string(payload->snapshot);
+ if (!channel_string) {
+ return NULL;
+ }
+ }
+
+ if (payload->blob) {
+ /* Optional data */
+ struct ast_json *to = ast_json_object_get(payload->blob, "to");
+ struct ast_json *from = ast_json_object_get(payload->blob, "from");
+ struct ast_json *rtt = ast_json_object_get(payload->blob, "rtt");
+ if (to) {
+ ast_str_append(&packet_string, 0, "To: %s\r\n", ast_json_string_get(to));
+ }
+ if (from) {
+ ast_str_append(&packet_string, 0, "From: %s\r\n", ast_json_string_get(from));
+ }
+ if (rtt) {
+ ast_str_append(&packet_string, 0, "RTT: %4.4f\r\n", ast_json_real_get(rtt));
+ }
+ }
+
+ ast_str_append(&packet_string, 0, "SSRC: 0x%.8x\r\n", ssrc);
+ ast_str_append(&packet_string, 0, "PT: %u(%s)\r\n", type, type== AST_RTP_RTCP_SR ? "SR" : "RR");
+ ast_str_append(&packet_string, 0, "ReportCount: %u\r\n", report_count);
+ if (type == AST_RTP_RTCP_SR) {
+ ast_str_append(&packet_string, 0, "SentNTP: %lu.%06lu\r\n",
+ (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec,
+ (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec);
+ ast_str_append(&packet_string, 0, "SentRTP: %u\r\n",
+ payload->report->sender_information.rtp_timestamp);
+ ast_str_append(&packet_string, 0, "SentPackets: %u\r\n",
+ payload->report->sender_information.packet_count);
+ ast_str_append(&packet_string, 0, "SentOctets: %u\r\n",
+ payload->report->sender_information.octet_count);
+ }
+
+ for (i = 0; i < report_count; i++) {
+ RAII_VAR(struct ast_str *, report_string, NULL, ast_free);
+
+ if (!payload->report->report_block[i]) {
+ break;
+ }
+
+ report_string = ast_str_create(256);
+ if (!report_string) {
+ return NULL;
+ }
+
+ ast_str_append(&report_string, 0, "Report%dSourceSSRC: 0x%.8x\r\n",
+ i, payload->report->report_block[i]->source_ssrc);
+ ast_str_append(&report_string, 0, "Report%dFractionLost: %d\r\n",
+ i, payload->report->report_block[i]->lost_count.fraction);
+ ast_str_append(&report_string, 0, "Report%dCumulativeLost: %u\r\n",
+ i, payload->report->report_block[i]->lost_count.packets);
+ ast_str_append(&report_string, 0, "Report%dHighestSequence: %u\r\n",
+ i, payload->report->report_block[i]->highest_seq_no & 0xffff);
+ ast_str_append(&report_string, 0, "Report%dSequenceNumberCycles: %u\r\n",
+ i, payload->report->report_block[i]->highest_seq_no >> 16);
+ ast_str_append(&report_string, 0, "Report%dIAJitter: %u\r\n",
+ i, payload->report->report_block[i]->ia_jitter);
+ ast_str_append(&report_string, 0, "Report%dLSR: %u\r\n",
+ i, payload->report->report_block[i]->lsr);
+ ast_str_append(&report_string, 0, "Report%dDLSR: %4.4f\r\n",
+ i, ((double)payload->report->report_block[i]->dlsr) / 65536);
+ ast_str_append(&packet_string, 0, "%s", ast_str_buffer(report_string));
+ }
+
+ return ast_manager_event_blob_create(EVENT_FLAG_REPORTING,
+ stasis_message_type(msg) == ast_rtp_rtcp_received_type() ? "RTCPReceived" : "RTCPSent",
+ "%s%s",
+ AS_OR(channel_string, ""),
+ ast_str_buffer(packet_string));
+}
+
+static struct ast_json *rtcp_report_to_json(struct stasis_message *msg,
+ const struct stasis_message_sanitizer *sanitize)
+{
+ struct rtcp_message_payload *payload = stasis_message_data(msg);
+ RAII_VAR(struct ast_json *, json_rtcp_report, NULL, ast_json_unref);
+ RAII_VAR(struct ast_json *, json_rtcp_report_blocks, NULL, ast_json_unref);
+ RAII_VAR(struct ast_json *, json_rtcp_sender_info, NULL, ast_json_unref);
+ RAII_VAR(struct ast_json *, json_channel, NULL, ast_json_unref);
+ int i;
+
+ json_rtcp_report_blocks = ast_json_array_create();
+ if (!json_rtcp_report_blocks) {
+ return NULL;
+ }
+
+ for (i = 0; i < payload->report->reception_report_count && payload->report->report_block[i]; i++) {
+ struct ast_json *json_report_block;
+ char str_lsr[32];
+ snprintf(str_lsr, sizeof(str_lsr), "%u", payload->report->report_block[i]->lsr);
+ json_report_block = ast_json_pack("{s: i, s: i, s: i, s: i, s: i, s: s, s: i}",
+ "source_ssrc", payload->report->report_block[i]->source_ssrc,
+ "fraction_lost", payload->report->report_block[i]->lost_count.fraction,
+ "packets_lost", payload->report->report_block[i]->lost_count.packets,
+ "highest_seq_no", payload->report->report_block[i]->highest_seq_no,
+ "ia_jitter", payload->report->report_block[i]->ia_jitter,
+ "lsr", str_lsr,
+ "dlsr", payload->report->report_block[i]->dlsr);
+ if (!json_report_block) {
+ return NULL;
+ }
+ if (ast_json_array_append(json_rtcp_report_blocks, json_report_block)) {
+ return NULL;
+ }
+ }
+
+ if (payload->report->type == AST_RTP_RTCP_SR) {
+ char sec[32];
+ char usec[32];
+ snprintf(sec, sizeof(sec), "%lu", (unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec);
+ snprintf(usec, sizeof(usec), "%lu", (unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec);
+ json_rtcp_sender_info = ast_json_pack("{s: s, s: s, s: i, s: i, s: i}",
+ "ntp_timestamp_sec", sec,
+ "ntp_timestamp_usec", usec,
+ "rtp_timestamp", payload->report->sender_information.rtp_timestamp,
+ "packets", payload->report->sender_information.packet_count,
+ "octets", payload->report->sender_information.octet_count);
+ if (!json_rtcp_sender_info) {
+ return NULL;
+ }
+ }
+
+ json_rtcp_report = ast_json_pack("{s: i, s: i, s: i, s: o, s: o}",
+ "ssrc", payload->report->ssrc,
+ "type", payload->report->type,
+ "report_count", payload->report->reception_report_count,
+ "sender_information", json_rtcp_sender_info ? ast_json_ref(json_rtcp_sender_info) : ast_json_ref(ast_json_null()),
+ "report_blocks", ast_json_ref(json_rtcp_report_blocks));
+ if (!json_rtcp_report) {
+ return NULL;
+ }
+
+ if (payload->snapshot) {
+ json_channel = ast_channel_snapshot_to_json(payload->snapshot, sanitize);
+ if (!json_channel) {
+ return NULL;
+ }
+ }
+
+ return ast_json_pack("{s: o, s: o, s: o}",
+ "channel", payload->snapshot ? ast_json_ref(json_channel) : ast_json_ref(ast_json_null()),
+ "rtcp_report", ast_json_ref(json_rtcp_report),
+ "blob", ast_json_deep_copy(payload->blob));
+}
+
+static void rtp_rtcp_report_dtor(void *obj)
+{
+ int i;
+ struct ast_rtp_rtcp_report *rtcp_report = obj;
+
+ for (i = 0; i < rtcp_report->reception_report_count; i++) {
+ ast_free(rtcp_report->report_block[i]);
+ }
+}
+
+struct ast_rtp_rtcp_report *ast_rtp_rtcp_report_alloc(unsigned int report_blocks)
+{
+ struct ast_rtp_rtcp_report *rtcp_report;
+
+ /* Size of object is sizeof the report + the number of report_blocks * sizeof pointer */
+ rtcp_report = ao2_alloc((sizeof(*rtcp_report) + report_blocks * sizeof(struct ast_rtp_rtcp_report_block *)),
+ rtp_rtcp_report_dtor);
+
+ return rtcp_report;
+}
+
+void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
+ struct stasis_message_type *message_type,
+ struct ast_rtp_rtcp_report *report,
+ struct ast_json *blob)
+{
+ RAII_VAR(struct rtcp_message_payload *, payload, NULL, ao2_cleanup);
+ RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
+
+ if (!message_type) {
+ return;
+ }
+
+ payload = ao2_alloc(sizeof(*payload), rtcp_message_payload_dtor);
+ if (!payload || !report) {
+ return;
+ }
+
+ if (!ast_strlen_zero(rtp->channel_uniqueid)) {
+ payload->snapshot = ast_channel_snapshot_get_latest(rtp->channel_uniqueid);
+ }
+ if (blob) {
+ payload->blob = blob;
+ ast_json_ref(blob);
+ }
+ ao2_ref(report, +1);
+ payload->report = report;
+
+ message = stasis_message_create(message_type, payload);
+ if (!message) {
+ return;
+ }
+
+ stasis_publish(ast_rtp_topic(), message);
+}
+
+/*!
+ * @{ \brief Define RTCP/RTP message types.
+ */
+STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_sent_type,
+ .to_ami = rtcp_report_to_ami,
+ .to_json = rtcp_report_to_json,);
+STASIS_MESSAGE_TYPE_DEFN(ast_rtp_rtcp_received_type,
+ .to_ami = rtcp_report_to_ami,
+ .to_json = rtcp_report_to_json,);
+/*! @} */
+
+struct stasis_topic *ast_rtp_topic(void)
+{
+ return rtp_topic;
+}
+
+static void rtp_engine_shutdown(void)
+{
+ int x;
+
+ ao2_cleanup(rtp_topic);
+ rtp_topic = NULL;
+ STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_received_type);
+ STASIS_MESSAGE_TYPE_CLEANUP(ast_rtp_rtcp_sent_type);
+
+ ast_rwlock_wrlock(&static_RTP_PT_lock);
+ for (x = 0; x < AST_RTP_MAX_PT; x++) {
+ ao2_cleanup(static_RTP_PT[x]);
+ static_RTP_PT[x] = NULL;
+ }
+ ast_rwlock_unlock(&static_RTP_PT_lock);
+
+ ast_rwlock_wrlock(&mime_types_lock);
+ for (x = 0; x < mime_types_len; x++) {
+ if (ast_rtp_mime_types[x].payload_type.format) {
+ rtp_engine_mime_type_cleanup(x);
+ }
+ }
+ mime_types_len = 0;
+ ast_rwlock_unlock(&mime_types_lock);
+}
+
+int ast_rtp_engine_init(void)
+{
ast_rwlock_init(&mime_types_lock);
ast_rwlock_init(&static_RTP_PT_lock);
+ rtp_topic = stasis_topic_create("rtp_topic");
+ if (!rtp_topic) {
+ return -1;
+ }
+ STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_sent_type);
+ STASIS_MESSAGE_TYPE_INIT(ast_rtp_rtcp_received_type);
+ ast_register_cleanup(rtp_engine_shutdown);
+
/* Define all the RTP mime types available */
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0, "audio", "G723", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0, "audio", "GSM", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "PCMU", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0, "audio", "G711U", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "PCMA", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0, "audio", "G711A", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0, "audio", "G726-32", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0, "audio", "DVI4", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0, "audio", "L16", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16", 16000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0, "audio", "L16-256", 16000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0, "audio", "LPC", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G729A", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0, "audio", "G.729", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0, "audio", "speex", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0, "audio", "speex", 16000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0, "audio", "speex", 32000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0, "audio", "iLBC", 8000);
+ set_next_mime_type(ast_format_g723, 0, "audio", "G723", 8000);
+ set_next_mime_type(ast_format_gsm, 0, "audio", "GSM", 8000);
+ set_next_mime_type(ast_format_ulaw, 0, "audio", "PCMU", 8000);
+ set_next_mime_type(ast_format_ulaw, 0, "audio", "G711U", 8000);
+ set_next_mime_type(ast_format_alaw, 0, "audio", "PCMA", 8000);
+ set_next_mime_type(ast_format_alaw, 0, "audio", "G711A", 8000);
+ set_next_mime_type(ast_format_g726, 0, "audio", "G726-32", 8000);
+ set_next_mime_type(ast_format_adpcm, 0, "audio", "DVI4", 8000);
+ set_next_mime_type(ast_format_slin, 0, "audio", "L16", 8000);
+ set_next_mime_type(ast_format_slin16, 0, "audio", "L16", 16000);
+ set_next_mime_type(ast_format_slin16, 0, "audio", "L16-256", 16000);
+ set_next_mime_type(ast_format_slin12, 0, "audio", "L16", 12000);
+ set_next_mime_type(ast_format_slin24, 0, "audio", "L16", 24000);
+ set_next_mime_type(ast_format_slin32, 0, "audio", "L16", 32000);
+ set_next_mime_type(ast_format_slin44, 0, "audio", "L16", 44000);
+ set_next_mime_type(ast_format_slin48, 0, "audio", "L16", 48000);
+ set_next_mime_type(ast_format_slin96, 0, "audio", "L16", 96000);
+ set_next_mime_type(ast_format_slin192, 0, "audio", "L16", 192000);
+ set_next_mime_type(ast_format_lpc10, 0, "audio", "LPC", 8000);
+ set_next_mime_type(ast_format_g729, 0, "audio", "G729", 8000);
+ set_next_mime_type(ast_format_g729, 0, "audio", "G729A", 8000);
+ set_next_mime_type(ast_format_g729, 0, "audio", "G.729", 8000);
+ set_next_mime_type(ast_format_speex, 0, "audio", "speex", 8000);
+ set_next_mime_type(ast_format_speex16, 0, "audio", "speex", 16000);
+ set_next_mime_type(ast_format_speex32, 0, "audio", "speex", 32000);
+ set_next_mime_type(ast_format_ilbc, 0, "audio", "iLBC", 8000);
/* this is the sample rate listed in the RTP profile for the G.722 codec, *NOT* the actual sample rate of the media stream */
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0, "audio", "G722", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0, "audio", "AAL2-G726-32", 8000);
+ set_next_mime_type(ast_format_g722, 0, "audio", "G722", 8000);
+ set_next_mime_type(ast_format_g726_aal2, 0, "audio", "AAL2-G726-32", 8000);
set_next_mime_type(NULL, AST_RTP_DTMF, "audio", "telephone-event", 8000);
set_next_mime_type(NULL, AST_RTP_CISCO_DTMF, "audio", "cisco-telephone-event", 8000);
set_next_mime_type(NULL, AST_RTP_CN, "audio", "CN", 8000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0, "video", "JPEG", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_PNG, 0), 0, "video", "PNG", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0, "video", "H261", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0, "video", "H263", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0, "video", "h263-1998", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0, "video", "H264", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0, "video", "MP4V-ES", 90000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0, "text", "RED", 1000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0, "text", "T140", 1000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
- set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
+ set_next_mime_type(ast_format_jpeg, 0, "video", "JPEG", 90000);
+ set_next_mime_type(ast_format_png, 0, "video", "PNG", 90000);
+ set_next_mime_type(ast_format_h261, 0, "video", "H261", 90000);
+ set_next_mime_type(ast_format_h263, 0, "video", "H263", 90000);
+ set_next_mime_type(ast_format_h263p, 0, "video", "h263-1998", 90000);
+ set_next_mime_type(ast_format_h264, 0, "video", "H264", 90000);
+ set_next_mime_type(ast_format_mp4, 0, "video", "MP4V-ES", 90000);
+ set_next_mime_type(ast_format_t140_red, 0, "text", "RED", 1000);
+ set_next_mime_type(ast_format_t140, 0, "text", "T140", 1000);
+ set_next_mime_type(ast_format_siren7, 0, "audio", "G7221", 16000);
+ set_next_mime_type(ast_format_siren14, 0, "audio", "G7221", 32000);
+ set_next_mime_type(ast_format_g719, 0, "audio", "G719", 48000);
+ /* Opus and VP8 */
+ set_next_mime_type(ast_format_opus, 0, "audio", "opus", 48000);
+ set_next_mime_type(ast_format_vp8, 0, "video", "VP8", 90000);
/* Define the static rtp payload mappings */
- add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
+ add_static_payload(0, ast_format_ulaw, 0);
#ifdef USE_DEPRECATED_G726
- add_static_payload(2, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
+ add_static_payload(2, ast_format_g726, 0);/* Technically this is G.721, but if Cisco can do it, so can we... */
#endif
- add_static_payload(3, ast_format_set(&tmpfmt, AST_FORMAT_GSM, 0), 0);
- add_static_payload(4, ast_format_set(&tmpfmt, AST_FORMAT_G723_1, 0), 0);
- add_static_payload(5, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0);/* 8 kHz */
- add_static_payload(6, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 16 kHz */
- add_static_payload(7, ast_format_set(&tmpfmt, AST_FORMAT_LPC10, 0), 0);
- add_static_payload(8, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0), 0);
- add_static_payload(9, ast_format_set(&tmpfmt, AST_FORMAT_G722, 0), 0);
- add_static_payload(10, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 2 channels */
- add_static_payload(11, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0), 0); /* 1 channel */
+ add_static_payload(3, ast_format_gsm, 0);
+ add_static_payload(4, ast_format_g723, 0);
+ add_static_payload(5, ast_format_adpcm, 0);/* 8 kHz */
+ add_static_payload(6, ast_format_adpcm, 0); /* 16 kHz */
+ add_static_payload(7, ast_format_lpc10, 0);
+ add_static_payload(8, ast_format_alaw, 0);
+ add_static_payload(9, ast_format_g722, 0);
+ add_static_payload(10, ast_format_slin, 0); /* 2 channels */
+ add_static_payload(11, ast_format_slin, 0); /* 1 channel */
add_static_payload(13, NULL, AST_RTP_CN);
- add_static_payload(16, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 11.025 kHz */
- add_static_payload(17, ast_format_set(&tmpfmt, AST_FORMAT_ADPCM, 0), 0); /* 22.050 kHz */
- add_static_payload(18, ast_format_set(&tmpfmt, AST_FORMAT_G729A, 0), 0);
+ add_static_payload(16, ast_format_adpcm, 0); /* 11.025 kHz */
+ add_static_payload(17, ast_format_adpcm, 0); /* 22.050 kHz */
+ add_static_payload(18, ast_format_g729, 0);
add_static_payload(19, NULL, AST_RTP_CN); /* Also used for CN */
- add_static_payload(26, ast_format_set(&tmpfmt, AST_FORMAT_JPEG, 0), 0);
- add_static_payload(31, ast_format_set(&tmpfmt, AST_FORMAT_H261, 0), 0);
- add_static_payload(34, ast_format_set(&tmpfmt, AST_FORMAT_H263, 0), 0);
- add_static_payload(97, ast_format_set(&tmpfmt, AST_FORMAT_ILBC, 0), 0);
- add_static_payload(98, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
- add_static_payload(99, ast_format_set(&tmpfmt, AST_FORMAT_H264, 0), 0);
+ add_static_payload(26, ast_format_jpeg, 0);
+ add_static_payload(31, ast_format_h261, 0);
+ add_static_payload(34, ast_format_h263, 0);
+ add_static_payload(97, ast_format_ilbc, 0);
+ add_static_payload(98, ast_format_h263p, 0);
+ add_static_payload(99, ast_format_h264, 0);
add_static_payload(101, NULL, AST_RTP_DTMF);
- add_static_payload(102, ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0);
- add_static_payload(103, ast_format_set(&tmpfmt, AST_FORMAT_H263_PLUS, 0), 0);
- add_static_payload(104, ast_format_set(&tmpfmt, AST_FORMAT_MP4_VIDEO, 0), 0);
- add_static_payload(105, ast_format_set(&tmpfmt, AST_FORMAT_T140RED, 0), 0); /* Real time text chat (with redundancy encoding) */
- add_static_payload(106, ast_format_set(&tmpfmt, AST_FORMAT_T140, 0), 0); /* Real time text chat */
- add_static_payload(110, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX, 0), 0);
- add_static_payload(111, ast_format_set(&tmpfmt, AST_FORMAT_G726, 0), 0);
- add_static_payload(112, ast_format_set(&tmpfmt, AST_FORMAT_G726_AAL2, 0), 0);
- add_static_payload(115, ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0);
- add_static_payload(116, ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0);
- add_static_payload(117, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX16, 0), 0);
- add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
- add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
+ add_static_payload(102, ast_format_siren7, 0);
+ add_static_payload(103, ast_format_h263p, 0);
+ add_static_payload(104, ast_format_mp4, 0);
+ add_static_payload(105, ast_format_t140_red, 0); /* Real time text chat (with redundancy encoding) */
+ add_static_payload(106, ast_format_t140, 0); /* Real time text chat */
+ add_static_payload(110, ast_format_speex, 0);
+ add_static_payload(111, ast_format_g726, 0);
+ add_static_payload(112, ast_format_g726_aal2, 0);
+ add_static_payload(115, ast_format_siren14, 0);
+ add_static_payload(116, ast_format_g719, 0);
+ add_static_payload(117, ast_format_speex16, 0);
+ add_static_payload(118, ast_format_slin16, 0); /* 16 Khz signed linear */
+ add_static_payload(119, ast_format_speex32, 0);
add_static_payload(121, NULL, AST_RTP_CISCO_DTMF); /* Must be type 121 */
+ add_static_payload(122, ast_format_slin12, 0);
+ add_static_payload(123, ast_format_slin24, 0);
+ add_static_payload(124, ast_format_slin32, 0);
+ add_static_payload(125, ast_format_slin44, 0);
+ add_static_payload(126, ast_format_slin48, 0);
+ add_static_payload(127, ast_format_slin96, 0);
+ /* payload types above 127 are not valid */
+ add_static_payload(96, ast_format_slin192, 0);
+ /* Opus and VP8 */
+ add_static_payload(100, ast_format_vp8, 0);
+ add_static_payload(107, ast_format_opus, 0);
return 0;
}
+
+time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp)
+{
+ return rtp->last_tx;
+}
+
+void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time)
+{
+ rtp->last_tx = time;
+}
+
+time_t ast_rtp_instance_get_last_rx(const struct ast_rtp_instance *rtp)
+{
+ return rtp->last_rx;
+}
+
+void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time)
+{
+ rtp->last_rx = time;
+}
projects / asterisk/asterisk.git / blobdiff