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codec negotiation: add incoming_call_offer_prefs option

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Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.

This patch does the following:

  Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.

  Adds a new ast_sip_session_caps structure that's set for each session media
object.

  Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.

ASTERISK-28756 #close

Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
This commit is contained in:
Kevin Harwell
2020-02-24 12:47:46 -06:00
parent 87fda066ea
commit 06dada3f01
11 changed files with 484 additions and 20 deletions
Download Patch File Download Diff File Expand all files Collapse all files

44
res/res_pjsip/pjsip_configuration.c
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@@ -1121,6 +1121,48 @@ 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,
struct ast_variable *var, void *obj)
{
struct ast_sip_endpoint *endpoint = obj;
unsigned int i;
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;
}
}
return -1;
}
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]);
}
return 0;
}
static void *sip_nat_hook_alloc(const char *name)
{
return ast_sorcery_generic_alloc(sizeof(struct ast_sip_nat_hook), NULL);
@@ -1966,6 +2008,8 @@ int ast_res_pjsip_initialize_configuration(void)
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "accept_multiple_sdp_answers", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.accept_multiple_sdp_answers));
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);
if (ast_sip_initialize_sorcery_transport()) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");