codec negotiation: add incoming_call_offer_prefs option

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

res/res_pjsip_session/pjsip_session_caps.c

@@ -0,0 +1,162 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2020, Sangoma Technologies Corporation
+ *
+ * Kevin Harwell <kharwell@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+#include "asterisk.h"
+
+#include "asterisk/astobj2.h"
+#include "asterisk/channel.h"
+#include "asterisk/format.h"
+#include "asterisk/format_cap.h"
+#include "asterisk/logger.h"
+#include "asterisk/sorcery.h"
+
+#include <pjsip_ua.h>
+
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+#include "asterisk/res_pjsip_session_caps.h"
+
+struct ast_sip_session_caps {
+	struct ast_format_cap *incoming_call_offer_cap;
+};
+
+static void log_caps(int level, const char *file, int line, const char *function,
+	const char *msg, const struct ast_sip_session *session,
+	const struct ast_sip_session_media *session_media, const struct ast_format_cap *local,
+	const struct ast_format_cap *remote, const struct ast_format_cap *joint)
+{
+	struct ast_str *s1;
+	struct ast_str *s2;
+	struct ast_str *s3;
+
+	if (level == __LOG_DEBUG && !DEBUG_ATLEAST(3)) {
+		return;
+	}
+
+	s1 = local ? ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN) : NULL;
+	s2 = remote ? ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN) : NULL;
+	s3 = joint ? ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN) : NULL;
+
+	ast_log(level, file, line, function, "'%s' %s '%s' capabilities -%s%s%s%s%s%s\n",
+		session->channel ? ast_channel_name(session->channel) :
+			ast_sorcery_object_get_id(session->endpoint),
+		msg ? msg : "-", ast_codec_media_type2str(session_media->type),
+		s1 ? " local: " : "", s1 ? ast_format_cap_get_names(local, &s1) : "",
+		s2 ? " remote: " : "", s2 ? ast_format_cap_get_names(remote, &s2) : "",
+		s3 ? " joint: " : "", s3 ? ast_format_cap_get_names(joint, &s3) : "");
+}
+
+static void sip_session_caps_destroy(void *obj)
+{
+	struct ast_sip_session_caps *caps = obj;
+
+	ao2_cleanup(caps->incoming_call_offer_cap);
+}
+
+struct ast_sip_session_caps *ast_sip_session_caps_alloc(void)
+{
+	return ao2_alloc_options(sizeof(struct ast_sip_session_caps),
+		sip_session_caps_destroy, AO2_ALLOC_OPT_LOCK_NOLOCK);
+}
+
+void ast_sip_session_set_incoming_call_offer_cap(struct ast_sip_session_caps *caps,
+	struct ast_format_cap *cap)
+{
+	ao2_cleanup(caps->incoming_call_offer_cap);
+	caps->incoming_call_offer_cap = ao2_bump(cap);
+}
+
+const struct ast_format_cap *ast_sip_session_get_incoming_call_offer_cap(
+	const struct ast_sip_session_caps *caps)
+{
+	return caps->incoming_call_offer_cap;
+}
+
+const struct ast_format_cap *ast_sip_session_join_incoming_call_offer_cap(
+	const struct ast_sip_session *session, const struct ast_sip_session_media *session_media,
+	const struct ast_format_cap *remote)
+{
+	enum ast_sip_call_codec_pref pref;
+	struct ast_format_cap *joint;
+	struct ast_format_cap *local;
+
+	joint = session_media->caps->incoming_call_offer_cap;
+
+	if (joint) {
+		/*
+		 * If the incoming call offer capabilities have been set elsewhere, e.g. dialplan
+		 * then those take precedence.
+		 */
+		return joint;
+	}
+
+	joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+	local = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+
+	if (!joint || !local) {
+		ast_log(LOG_ERROR, "Failed to allocate %s incoming call offer capabilities\n",
+				ast_codec_media_type2str(session_media->type));
+
+		ao2_cleanup(joint);
+		ao2_cleanup(local);
+		return NULL;
+	}
+
+	pref = session->endpoint->media.incoming_call_offer_pref;
+	ast_format_cap_append_from_cap(local, session->endpoint->media.codecs,
+		session_media->type);
+
+	if (pref < AST_SIP_CALL_CODEC_PREF_REMOTE) {
+		ast_format_cap_get_compatible(local, remote, joint); /* Prefer local */
+	} else {
+		ast_format_cap_get_compatible(remote, local, joint); /* Prefer remote */
+	}
+
+	if (ast_format_cap_empty(joint)) {
+		log_caps(LOG_NOTICE, "No joint incoming", session, session_media, local, remote, NULL);
+
+		ao2_ref(joint, -1);
+		ao2_ref(local, -1);
+		return NULL;
+	}
+
+	if (pref == AST_SIP_CALL_CODEC_PREF_LOCAL_SINGLE ||
+		pref == AST_SIP_CALL_CODEC_PREF_REMOTE_SINGLE ||
+		session->endpoint->preferred_codec_only) {
+
+		/*
+		 * Save the most preferred one. Session capabilities are per stream and
+		 * a stream only carries a single media type, so no reason to worry with
+		 * the type here (i.e different or multiple types)
+		 */
+		struct ast_format *single = ast_format_cap_get_format(joint, 0);
+		/* Remove all formats */
+		ast_format_cap_remove_by_type(joint, AST_MEDIA_TYPE_UNKNOWN);
+		/* Put the most preferred one back */
+		ast_format_cap_append(joint, single, 0);
+		ao2_ref(single, -1);
+	}
+
+	log_caps(LOG_DEBUG, "Joint incoming", session, session_media, local, remote, joint);
+
+	ao2_ref(local, -1);
+
+	ast_sip_session_set_incoming_call_offer_cap(session_media->caps, joint);
+
+	return joint;
+}