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codec_negotiation: Implement outgoing_call_offer_pref

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Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.

* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)

* Add "call_direction" to res_pjsip_session.

* Update pjsip_session_caps.c to make the functions more generic
  so they could be used for both incoming and outgoing.

* Update ast_sip_session_create_outgoing to create the
  pending_media_state->topology with the results of
  ast_sip_session_create_joint_call_stream().

* The endpoint "preferred_codec_only" option now automatically sets
  AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.

* A helper function ast_stream_get_format_count() was added to
  streams to return the current count of formats.

ASTERISK-28777

Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
This commit is contained in:
George Joseph
2020-03-13 13:40:46 -06:00
committed by Friendly Automation
parent 57a457c26c
commit 2ee455958e
13 changed files with 408 additions and 297 deletions
Download Patch File Download Diff File Expand all files Collapse all files

78
res/res_pjsip/pjsip_configuration.c
View File

@@ -1121,43 +1121,57 @@ static int contact_user_to_str(const void *obj, const intptr_t *args, char **buf
return 0;
}
static const char *sip_call_codec_pref_strings[] = {
[AST_SIP_CALL_CODEC_PREF_LOCAL] = "local",
[AST_SIP_CALL_CODEC_PREF_LOCAL_LIMIT] = "local_limit",
[AST_SIP_CALL_CODEC_PREF_LOCAL_SINGLE] = "local_single",
[AST_SIP_CALL_CODEC_PREF_REMOTE] = "remote",
[AST_SIP_CALL_CODEC_PREF_REMOTE_LIMIT] = "remote_limit",
[AST_SIP_CALL_CODEC_PREF_REMOTE_SINGLE] = "remote_single",
};
static int incoming_call_offer_pref_handler(const struct aco_option *opt,
static int call_offer_pref_handler(const struct aco_option *opt,
struct ast_variable *var, void *obj)
{
struct ast_sip_endpoint *endpoint = obj;
unsigned int i;
struct ast_flags pref = { 0, };
int outgoing = strcmp(var->name, "outgoing_call_offer_pref") == 0;
for (i = 0; i < ARRAY_LEN(sip_call_codec_pref_strings); ++i) {
if (!strcmp(var->value, sip_call_codec_pref_strings[i])) {
/* Local and remote limit are not available values for this option */
if (i == AST_SIP_CALL_CODEC_PREF_LOCAL_LIMIT ||
i == AST_SIP_CALL_CODEC_PREF_REMOTE_LIMIT) {
return -1;
}
endpoint->media.incoming_call_offer_pref = i;
return 0;
}
if (strcmp(var->value, "local") == 0) {
ast_set_flag(&pref, AST_SIP_CALL_CODEC_PREF_LOCAL | AST_SIP_CALL_CODEC_PREF_INTERSECT | AST_SIP_CALL_CODEC_PREF_ALL);
} else if (outgoing && strcmp(var->value, "local_merge") == 0) {
ast_set_flag(&pref, AST_SIP_CALL_CODEC_PREF_LOCAL | AST_SIP_CALL_CODEC_PREF_UNION | AST_SIP_CALL_CODEC_PREF_ALL);
} else if (strcmp(var->value, "local_first") == 0) {
ast_set_flag(&pref, AST_SIP_CALL_CODEC_PREF_LOCAL | AST_SIP_CALL_CODEC_PREF_INTERSECT | AST_SIP_CALL_CODEC_PREF_FIRST);
} else if (strcmp(var->value, "remote") == 0) {
ast_set_flag(&pref, AST_SIP_CALL_CODEC_PREF_REMOTE | AST_SIP_CALL_CODEC_PREF_INTERSECT | AST_SIP_CALL_CODEC_PREF_ALL);
} else if (outgoing && strcmp(var->value, "remote_merge") == 0) {
ast_set_flag(&pref, AST_SIP_CALL_CODEC_PREF_REMOTE | AST_SIP_CALL_CODEC_PREF_UNION | AST_SIP_CALL_CODEC_PREF_ALL);
} else if (strcmp(var->value, "remote_first") == 0) {
ast_set_flag(&pref, AST_SIP_CALL_CODEC_PREF_REMOTE | AST_SIP_CALL_CODEC_PREF_UNION | AST_SIP_CALL_CODEC_PREF_FIRST);
} else {
return -1;
}
return -1;
if (outgoing) {
endpoint->media.outgoing_call_offer_pref = pref;
} else {
endpoint->media.incoming_call_offer_pref = pref;
}
return 0;
}
static int incoming_call_offer_pref_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct ast_sip_endpoint *endpoint = obj;
if (ARRAY_IN_BOUNDS(endpoint->media.incoming_call_offer_pref, sip_call_codec_pref_strings)) {
*buf = ast_strdup(sip_call_codec_pref_strings[endpoint->media.incoming_call_offer_pref]);
*buf = ast_strdup(ast_sip_call_codec_pref_to_str(endpoint->media.incoming_call_offer_pref));
if (!(*buf)) {
return -1;
}
return 0;
}
static int outgoing_call_offer_pref_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct ast_sip_endpoint *endpoint = obj;
*buf = ast_strdup(ast_sip_call_codec_pref_to_str(endpoint->media.outgoing_call_offer_pref));
if (!(*buf)) {
return -1;
}
return 0;
@@ -1345,6 +1359,16 @@ static int sip_endpoint_apply_handler(const struct ast_sorcery *sorcery, void *o
return -1;
}
if (endpoint->preferred_codec_only) {
if (endpoint->media.incoming_call_offer_pref.flags != (AST_SIP_CALL_CODEC_PREF_LOCAL | AST_SIP_CALL_CODEC_PREF_INTERSECT | AST_SIP_CALL_CODEC_PREF_ALL)) {
ast_log(LOG_ERROR, "Setting both preferred_codec_only and incoming_call_offer_pref is not supported on endpoint '%s'\n",
ast_sorcery_object_get_id(endpoint));
return -1;
}
ast_clear_flag(&endpoint->media.incoming_call_offer_pref, AST_SIP_CALL_CODEC_PREF_ALL);
ast_set_flag(&endpoint->media.incoming_call_offer_pref, AST_SIP_CALL_CODEC_PREF_FIRST);
}
endpoint->media.topology = ast_stream_topology_create_from_format_cap(endpoint->media.codecs);
if (!endpoint->media.topology) {
return -1;
@@ -2009,7 +2033,9 @@ int ast_res_pjsip_initialize_configuration(void)
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "suppress_q850_reason_headers", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, suppress_q850_reason_headers));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "ignore_183_without_sdp", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, ignore_183_without_sdp));
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "incoming_call_offer_pref", "local",
incoming_call_offer_pref_handler, incoming_call_offer_pref_to_str, NULL, 0, 0);
call_offer_pref_handler, incoming_call_offer_pref_to_str, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "outgoing_call_offer_pref", "remote",
call_offer_pref_handler, outgoing_call_offer_pref_to_str, NULL, 0, 0);
if (ast_sip_initialize_sorcery_transport()) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");