[Asterisk-code-review] chan pjsip: Multistream: Use underlying multistream structures. (asterisk[master])
Mark Michelson
asteriskteam at digium.com
Mon Jun 5 13:24:55 CDT 2017
Mark Michelson has uploaded a new change for review. ( https://gerrit.asterisk.org/5760 )
Change subject: chan_pjsip: Multistream: Use underlying multistream structures.
......................................................................
chan_pjsip: Multistream: Use underlying multistream structures.
This commit is step one towards making chan_pjsip capable of supporting
multiple streams of a given type. The basic thing that's been done here
is to make it so that rather than using ast_format_cap structures in
places, we instead are using ast_stream_topology and ast_stream. There
is no functionality change being made by this commit; you still can only
have a single audio and single video stream on a call with a PJSIP
endpoint.
Throughout this commit some XXX comments have been added in places where
some assumptions may no longer hold when actually making the code
support multistream.
Change-Id: I0358ea3108e2c06ae43308abddf216a23deccb90
---
M channels/chan_pjsip.c
M channels/pjsip/dialplan_functions.c
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
M include/asterisk/stream.h
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
8 files changed, 244 insertions(+), 78 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/60/5760/1
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 5bf339e..41c12ee 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -64,6 +64,7 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
+#include "asterisk/stream.h"
#include "pjsip/include/chan_pjsip.h"
#include "pjsip/include/dialplan_functions.h"
@@ -430,6 +431,37 @@
}
}
+/*!
+ * \brief Determine if topologies have compatible formats
+ *
+ * This will return true if ANY formats in the topology are compatible with one another.
+ * It does not guarantee that each individual stream is compatible with the other streams.
+ *
+ * XXX When supporting true multistream, we will need to be sure to mark which streams from
+ * top1 are compatible with which streams from top2. Then the ones that are not compatible
+ * will need to be marked as "removed" so that they are negotiated as expected.
+ *
+ * \param top1 Topology 1
+ * \param top2 Topology 2
+ * \retval 1 The topologies have at least one compatible format
+ * \retval 0 The topologies have no compatible formats or an error occurred.
+ */
+static int compatible_formats_exist(struct ast_stream_topology *top1, struct ast_stream_topology *top2)
+{
+ int i;
+ struct ast_format_cap *cap1;
+ struct ast_format_cap *cap2;
+
+ cap1 = ast_format_cap_from_stream_topology(top1);
+ cap2 = ast_format_cap_from_stream_topology(top2);
+
+ if (!cap1 || !cap2) {
+ return 0;
+ }
+
+ return ast_format_cap_iscompatible(cap1, cap2);
+}
+
/*! \brief Function called to create a new PJSIP Asterisk channel */
static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
{
@@ -440,10 +472,6 @@
struct ast_variable *var;
if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
- return NULL;
- }
- caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (!caps) {
return NULL;
}
@@ -457,14 +485,12 @@
ast_sorcery_object_get_id(session->endpoint),
(unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
if (!chan) {
- ao2_ref(caps, -1);
return NULL;
}
ast_channel_tech_set(chan, &chan_pjsip_tech);
if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
- ao2_ref(caps, -1);
ast_channel_unlock(chan);
ast_hangup(chan);
return NULL;
@@ -474,11 +500,17 @@
ast_channel_tech_pvt_set(chan, channel);
- if (!ast_format_cap_count(session->req_caps) ||
- !ast_format_cap_iscompatible(session->req_caps, session->endpoint->media.codecs)) {
- ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
+ if (!ast_stream_topology_get_count(session->req_topology) ||
+ !compatible_formats_exist(session->req_topology, session->endpoint->media.codecs)) {
+ caps = ast_format_cap_from_stream_topology(session->endpoint->media.codecs);
} else {
- ast_format_cap_append_from_cap(caps, session->req_caps, AST_MEDIA_TYPE_UNKNOWN);
+ caps = ast_format_cap_from_stream_topology(session->req_topology);
+ }
+
+ if (!caps) {
+ ast_channel_unlock(chan);
+ ast_hangup(chan);
+ return NULL;
}
ast_channel_nativeformats_set(chan, caps);
@@ -2105,7 +2137,7 @@
struct request_data {
struct ast_sip_session *session;
- struct ast_format_cap *caps;
+ struct ast_stream_topology *topology;
const char *dest;
int cause;
};
@@ -2179,7 +2211,7 @@
}
}
- if (!(session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user, req_data->caps))) {
+ if (!(session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user, req_data->topology))) {
ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name);
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
return -1;
@@ -2196,8 +2228,12 @@
struct request_data req_data;
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
- req_data.caps = cap;
+ req_data.topology = ast_stream_topology_create_from_format_cap(cap);
req_data.dest = data;
+
+ if (!req_data.topology) {
+ return NULL;
+ }
if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
*cause = req_data.cause;
@@ -2205,6 +2241,7 @@
}
session = req_data.session;
+ ast_stream_topology_free(req_data.topology);
if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
/* Session needs to be terminated prematurely */
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
index 332df70..dd4cca8 100644
--- a/channels/pjsip/dialplan_functions.c
+++ b/channels/pjsip/dialplan_functions.c
@@ -441,6 +441,7 @@
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
+#include "asterisk/stream.h"
#include "include/chan_pjsip.h"
#include "include/dialplan_functions.h"
@@ -924,26 +925,20 @@
return 0;
}
-static int media_offer_read_av(struct ast_sip_session *session, char *buf,
- size_t len, enum ast_media_type media_type)
+static void append_format_names(struct ast_format_cap *caps, char *buf, size_t *len)
{
int i, size = 0;
- for (i = 0; i < ast_format_cap_count(session->req_caps); i++) {
- struct ast_format *fmt = ast_format_cap_get_format(session->req_caps, i);
-
- if (ast_format_get_type(fmt) != media_type) {
- ao2_ref(fmt, -1);
- continue;
- }
+ for (i = 0; i < ast_format_cap_count(caps); i++) {
+ struct ast_format *fmt = ast_format_cap_get_format(caps, i);
/* add one since we'll include a comma */
size = strlen(ast_format_get_name(fmt)) + 1;
- if (len < size) {
+ if (*len < size) {
ao2_ref(fmt, -1);
break;
}
- len -= size;
+ *len -= size;
/* no reason to use strncat here since we have already ensured buf has
enough space, so strcat can be safely used */
@@ -952,8 +947,29 @@
ao2_ref(fmt, -1);
}
+}
- if (size) {
+static int media_offer_read_av(struct ast_sip_session *session, char *buf,
+ size_t len, enum ast_media_type media_type)
+{
+ int i;
+
+ for (i = 0; i < ast_stream_topology_get_count(session->req_topology); ++i) {
+ struct ast_stream *stream;
+ struct ast_format_cap *caps;
+
+ stream = ast_stream_topology_get_stream(session->req_topology, i);
+
+ if (ast_stream_get_type(stream) != media_type) {
+ continue;
+ }
+
+ caps = ast_stream_get_formats(stream);
+
+ append_format_names(caps, buf, &len);
+ }
+
+ if (!ast_strlen_zero(buf)) {
/* remove the extra comma */
buf[strlen(buf) - 1] = '\0';
}
@@ -969,9 +985,22 @@
static int media_offer_write_av(void *obj)
{
struct media_offer_data *data = obj;
+ struct ast_stream *stream;
+ struct ast_format_cap *caps;
- ast_format_cap_remove_by_type(data->session->req_caps, data->media_type);
- ast_format_cap_update_by_allow_disallow(data->session->req_caps, data->value, 1);
+ /* XXX This method won't work when it comes time to do multistream support. The proper way to do this
+ * will either be to
+ * a) Alter all media streams of a particular type.
+ * b) Change the dialplan function to be able to specify which stream to alter and alter only that
+ * one stream
+ */
+ stream = ast_stream_topology_get_first_stream_by_type(data->session->req_topology, data->media_type);
+ if (!stream) {
+ return 0;
+ }
+ caps = ast_stream_get_formats(stream);
+ ast_format_cap_remove_by_type(caps, data->media_type);
+ ast_format_cap_update_by_allow_disallow(caps, data->value, 1);
return 0;
}
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 59f1710..04a2b65 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -665,7 +665,7 @@
/*! T.38 (FoIP) options */
struct ast_sip_t38_configuration t38;
/*! Configured codecs */
- struct ast_format_cap *codecs;
+ struct ast_stream_topology *codecs;
/*! DSCP TOS bits for audio streams */
unsigned int tos_audio;
/*! Priority for audio streams */
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 10e55f1..9bffa16 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -140,7 +140,7 @@
/*! Identity of endpoint this session deals with */
struct ast_party_id id;
/*! Requested capabilities */
- struct ast_format_cap *req_caps;
+ struct ast_stream_topology *req_topology;
/*! Optional DSP, used only for inband DTMF/Fax-CNG detection if configured */
struct ast_dsp *dsp;
/*! Whether the termination of the session should be deferred */
@@ -448,11 +448,11 @@
* \param contact The contact that this session will communicate with
* \param location Name of the location to call, be it named location or explicit URI. Overrides contact if present.
* \param request_user Optional request user to place in the request URI if permitted
- * \param req_caps The requested capabilities
+ * \param req_topology The requested capabilities
*/
struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint,
struct ast_sip_contact *contact, const char *location, const char *request_user,
- struct ast_format_cap *req_caps);
+ struct ast_stream_topology *req_topology);
/*!
* \brief Terminate a session and, if possible, send the provided response code
diff --git a/include/asterisk/stream.h b/include/asterisk/stream.h
index b453ab9..cf60d42 100644
--- a/include/asterisk/stream.h
+++ b/include/asterisk/stream.h
@@ -371,7 +371,7 @@
* since a new format capabilities structure is created for each media type.
*
* \note Each stream will have its name set to the corresponding media type.
- * For example: "AST_MEDIA_TYPE_AUDIO".
+ * For example: "audio".
*
* \note Each stream will be set to the sendrecv state.
*
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 679c883..c8a9fde 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -22,6 +22,7 @@
#include "asterisk/test.h"
#include "asterisk/statsd.h"
#include "asterisk/pbx.h"
+#include "asterisk/stream.h"
/*! \brief Number of buckets for persistent endpoint information */
#define PERSISTENT_BUCKETS 53
@@ -359,6 +360,59 @@
}
*buf = ast_strdup(*buf);
+ return 0;
+}
+
+static int codec_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
+{
+ struct ast_sip_endpoint *endpoint = obj;
+ struct ast_format_cap *cap;
+ struct ast_stream_topology *topology;
+
+ cap = ast_format_cap_from_stream_topology(endpoint->media.codecs);
+ if (!cap) {
+ return -1;
+ }
+ if (ast_format_cap_update_by_allow_disallow(cap, var->value, aco_option_get_flags(opt))) {
+ ao2_ref(cap, -1);
+ return -1;
+ }
+ /* XXX This topology creation may not work as-is for multistream.
+ * The problem is that the conversion from format_cap to topology
+ * will only create a stream per media type. If the topology should
+ * have multiple audio/video streams, then a different method
+ * will need to be used
+ */
+ topology = ast_stream_topology_create_from_format_cap(cap);
+ if (!topology) {
+ ao2_ref(cap, -1);
+ return -1;
+ }
+ ast_stream_topology_free(endpoint->media.codecs);
+ endpoint->media.codecs = topology;
+ ao2_ref(cap, -1);
+ return 0;
+}
+
+
+static int codec_to_str(const void *obj, const intptr_t *args, char **buf)
+{
+ const struct ast_sip_endpoint *endpoint;
+ struct ast_str *codecs;
+ struct ast_format_cap *cap;
+
+ endpoint = obj;
+ codecs = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
+
+ cap = ast_format_cap_from_stream_topology(endpoint->media.codecs);
+ if (!cap) {
+ return -1;
+ }
+
+ *buf = ast_strdup(ast_format_cap_get_names(cap, &codecs));
+ if (!*buf) {
+ return -1;
+ }
return 0;
}
@@ -1831,8 +1885,8 @@
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "type", "", OPT_NOOP_T, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "context", "default", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, context));
- ast_sorcery_object_field_register_alias(sip_sorcery, "endpoint", "disallow", "", OPT_CODEC_T, 0, FLDSET(struct ast_sip_endpoint, media.codecs));
- ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow", "", OPT_CODEC_T, 1, FLDSET(struct ast_sip_endpoint, media.codecs));
+ ast_sorcery_object_field_register_custom_alias(sip_sorcery, "endpoint", "disallow", "", codec_handler, codec_to_str, NULL, 0, 0);
+ ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "allow", "", codec_handler, codec_to_str, NULL, 1, 0);
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtmf_mode", "rfc4733", dtmf_handler, dtmf_to_str, NULL, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtp_ipv6", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.ipv6));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtp_symmetric", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.symmetric));
@@ -2059,7 +2113,7 @@
ast_string_field_free_memory(endpoint);
- ao2_ref(endpoint->media.codecs, -1);
+ ast_stream_topology_free(endpoint->media.codecs);
subscription_configuration_destroy(&endpoint->subscription);
info_configuration_destroy(&endpoint->info);
media_configuration_destroy(&endpoint->media);
@@ -2101,7 +2155,7 @@
ao2_cleanup(endpoint);
return NULL;
}
- if (!(endpoint->media.codecs = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+ if (!(endpoint->media.codecs = ast_stream_topology_alloc())) {
ao2_cleanup(endpoint);
return NULL;
}
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 97e365c..49f2f5e 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -51,6 +51,7 @@
#include "asterisk/sdp_srtp.h"
#include "asterisk/dsp.h"
#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT */
+#include "asterisk/stream.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
@@ -91,18 +92,6 @@
case AST_MEDIA_TYPE_END: break;
}
return -1;
-}
-
-/*! \brief Remove all other cap types but the one given */
-static void format_cap_only_type(struct ast_format_cap *caps, enum ast_media_type media_type)
-{
- int i = 0;
- while (i <= AST_MEDIA_TYPE_TEXT) {
- if (i != media_type && i != AST_MEDIA_TYPE_UNKNOWN) {
- ast_format_cap_remove_by_type(caps, i);
- }
- i += 1;
- }
}
static int send_keepalive(const void *data)
@@ -358,6 +347,10 @@
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
int dsp_features = 0;
+ struct ast_stream *req_stream;
+ struct ast_format_cap *req_caps;
+ struct ast_stream *endpoint_stream;
+ struct ast_format_cap *endpoint_caps;
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
@@ -366,12 +359,17 @@
return -1;
}
+ endpoint_stream = ast_stream_topology_get_first_stream_by_type(session->endpoint->media.codecs, media_type);
+ if (!endpoint_stream) {
+ return -1;
+ }
+ endpoint_caps = ast_stream_get_formats(endpoint_stream);
+
/* get the endpoint capabilities */
if (direct_media_enabled) {
- ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
- format_cap_only_type(caps, media_type);
+ ast_format_cap_get_compatible(endpoint_caps, session->direct_media_cap, caps);
} else {
- ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
+ ast_format_cap_append_from_cap(caps, endpoint_caps, media_type);
}
/* get the capabilities on the peer */
@@ -402,7 +400,18 @@
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
- ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
+ req_stream = ast_stream_topology_get_first_stream_by_type(session->req_topology, media_type);
+ if (!req_stream) {
+ req_stream = ast_stream_alloc(ast_codec_media_type2str(media_type), media_type);
+ if (!req_stream) {
+ return -1;
+ }
+ ast_stream_set_formats(req_stream, joint);
+ ast_stream_topology_append_stream(session->req_topology, req_stream);
+ } else {
+ req_caps = ast_stream_get_formats(req_stream);
+ ast_format_cap_replace_from_cap(req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
+ }
if (session->channel) {
ast_channel_lock(session->channel);
@@ -968,7 +977,7 @@
}
/* If no type formats have been configured reject this stream */
- if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
+ if (!ast_stream_topology_get_first_stream_by_type(session->endpoint->media.codecs, media_type)) {
ast_debug(3, "Endpoint has no codecs for media type '%s', declining stream\n", session_media->stream_type);
return 0;
}
@@ -1165,18 +1174,26 @@
int rtp_code;
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
- int use_override_prefs = ast_format_cap_count(session->req_caps);
+ int use_override_prefs = ast_stream_topology_get_count(session->req_topology);
+ /* XXX Getting first stream by media type is not going to cut it when we're creating multiple streams of the same type */
+ struct ast_stream *req_stream = ast_stream_topology_get_first_stream_by_type(session->req_topology, media_type);
+ struct ast_stream *endpoint_config_stream = ast_stream_topology_get_first_stream_by_type(session->endpoint->media.codecs, media_type);
+ struct ast_format_cap *req_caps;
+ struct ast_format_cap *endpoint_config_caps;
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
- if ((use_override_prefs && !ast_format_cap_has_type(session->req_caps, media_type)) ||
- (!use_override_prefs && !ast_format_cap_has_type(session->endpoint->media.codecs, media_type))) {
+ if ((use_override_prefs && !req_stream) ||
+ (!use_override_prefs && !endpoint_config_stream)) {
/* If no type formats are configured don't add a stream */
return 0;
} else if (!session_media->rtp && create_rtp(session, session_media)) {
return -1;
}
+
+ req_caps = req_stream ? ast_stream_get_formats(req_stream) : NULL;
+ endpoint_config_caps = ast_stream_get_formats(endpoint_config_stream);
set_ice_components(session, session_media);
enable_rtcp(session, session_media, NULL);
@@ -1244,12 +1261,12 @@
}
if (direct_media_enabled) {
- ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
- } else if (!ast_format_cap_count(session->req_caps) ||
- !ast_format_cap_iscompatible(session->req_caps, session->endpoint->media.codecs)) {
- ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
+ ast_format_cap_get_compatible(endpoint_config_caps, session->direct_media_cap, caps);
+ } else if (!req_caps || !ast_format_cap_count(req_caps) ||
+ !ast_format_cap_iscompatible(req_caps, endpoint_config_caps)) {
+ ast_format_cap_append_from_cap(caps, endpoint_config_caps, media_type);
} else {
- ast_format_cap_append_from_cap(caps, session->req_caps, media_type);
+ ast_format_cap_append_from_cap(caps, req_caps, media_type);
}
for (index = 0; index < ast_format_cap_count(caps); ++index) {
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 9e3abf2..df5593d 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -47,6 +47,7 @@
#include "asterisk/features_config.h"
#include "asterisk/pickup.h"
#include "asterisk/test.h"
+#include "asterisk/stream.h"
#define SDP_HANDLER_BUCKETS 11
@@ -1330,7 +1331,7 @@
ao2_cleanup(session->endpoint);
ao2_cleanup(session->aor);
ao2_cleanup(session->contact);
- ao2_cleanup(session->req_caps);
+ ast_stream_topology_free(session->req_topology);
ao2_cleanup(session->direct_media_cap);
ast_dsp_free(session->dsp);
@@ -1422,8 +1423,8 @@
if (!session->direct_media_cap) {
return NULL;
}
- session->req_caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
- if (!session->req_caps) {
+ session->req_topology = ast_stream_topology_alloc();
+ if (!session->req_topology) {
return NULL;
}
session->datastores = ao2_container_alloc(DATASTORE_BUCKETS, datastore_hash, datastore_cmp);
@@ -1704,7 +1705,7 @@
struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint,
struct ast_sip_contact *contact, const char *location, const char *request_user,
- struct ast_format_cap *req_caps)
+ struct ast_stream_topology *req_topology)
{
const char *uri = NULL;
RAII_VAR(struct ast_sip_aor *, found_aor, NULL, ao2_cleanup);
@@ -1767,22 +1768,50 @@
session->aor = ao2_bump(found_aor);
ast_party_id_copy(&session->id, &endpoint->id.self);
- if (ast_format_cap_count(req_caps)) {
- /* get joint caps between req_caps and endpoint caps */
- struct ast_format_cap *joint_caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (ast_stream_topology_get_count(req_topology) > 0) {
+ /* get joint caps between req_topology and endpoint topology */
+ int i;
- ast_format_cap_get_compatible(req_caps, endpoint->media.codecs, joint_caps);
+ for (i = 0; i < ast_stream_topology_get_count(req_topology); ++i) {
+ struct ast_stream *req_stream;
+ struct ast_stream *config_stream;
+ struct ast_format_cap *req_cap;
+ struct ast_format_cap *config_cap;
+ struct ast_format_cap *joint_cap;
+ enum ast_media_type type;
- /* if joint caps */
- if (ast_format_cap_count(joint_caps)) {
- /* copy endpoint caps into session->req_caps */
- ast_format_cap_append_from_cap(session->req_caps,
- endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
- /* replace instances of joint caps equivalents in session->req_caps */
- ast_format_cap_replace_from_cap(session->req_caps, joint_caps,
- AST_MEDIA_TYPE_UNKNOWN);
+ req_stream = ast_stream_topology_get_stream(req_topology, i);
+ type = ast_stream_get_type(req_stream);
+ /* XXX The retrieval of config_stream here is currently assuming that there
+ * is but one stream of a given media type. This may need to change when
+ * supporting true multistream mode. If all streams support the same codecs,
+ * then this will still work though.
+ */
+ config_stream = ast_stream_topology_get_first_stream_by_type(endpoint->media.codecs, type);
+
+ if (!config_stream) {
+ continue;
+ }
+
+ req_cap = ast_stream_get_formats(req_stream);
+ config_cap = ast_stream_get_formats(config_stream);
+
+ joint_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (!joint_cap) {
+ continue;
+ }
+
+ ast_format_cap_get_compatible(req_cap, config_cap, joint_cap);
+ if (ast_format_cap_count(joint_cap) > 0) {
+ /* copy endpoint caps into req_caps */
+ ast_format_cap_append_from_cap(req_cap,
+ config_cap, AST_MEDIA_TYPE_UNKNOWN);
+ /* replace instances of joint caps equivalents in req_caps */
+ ast_format_cap_replace_from_cap(req_cap, joint_cap,
+ AST_MEDIA_TYPE_UNKNOWN);
+ ast_stream_topology_append_stream(session->req_topology, ast_stream_clone(req_stream));
+ }
}
- ao2_cleanup(joint_caps);
}
if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
--
To view, visit https://gerrit.asterisk.org/5760
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I0358ea3108e2c06ae43308abddf216a23deccb90
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
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