mirror of http://gerrit.asterisk.org/asterisk
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: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76fchanges/42/13842/3
Kevin Harwell
5 years ago
parent
87fda066ea
commit
06dada3f01
@ -0,0 +1,53 @@
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Subject: res_pjsip
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Subject: res_pjsip_session
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Master-Only: True
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A new option, incoming_call_offer_pref, was added to res_pjsip endpoints that
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specifies the preferred order of codecs to use between those received in the
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offer, and those set in the configuration.
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Valid values include:
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local - prefer and order by configuration (default).
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local_single - prefer and order by configuration, but only choose 'top'
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most codec
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remote - prefer and order by incoming sdp.
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remote_single - prefer and order by incoming sdp, but only choose 'top' most
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most codec
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Example A:
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[alice]
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type=endpoint
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incoming_call_offer_pref=local
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allow=!all,opus,alaw,ulaw
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Alice's incoming sdp=g722,ulaw,alaw
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RESULT: alaw,ulaw
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Example B:
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[alice]
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type=endpoint
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incoming_call_offer_pref=local_single
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allow=!all,opus,alaw,ulaw
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Alice's incoming sdp=g722,ulaw,alaw
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RESULT: alaw
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Example C:
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[alice]
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type=endpoint
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incoming_call_offer_pref=remote
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allow=!all,opus,alaw,ulaw
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Alice's incoming sdp=g722,ulaw,alaw
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RESULT: ulaw,alaw
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Example D:
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[alice]
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type=endpoint
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incoming_call_offer_pref=remote_single
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allow=!all,opus,alaw,ulaw
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Alice's incoming sdp=g722,ulaw,alaw
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RESULT: ulaw
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@ -0,0 +1,82 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2020, Sangoma Technologies Corporation
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*
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* Kevin Harwell <kharwell@sangoma.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#ifndef RES_PJSIP_SESSION_CAPS_H
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#define RES_PJSIP_SESSION_CAPS_H
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struct ast_format_cap;
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struct ast_sip_session;
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struct ast_sip_session_media;
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struct ast_sip_session_caps;
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/*!
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* \brief Allocate a SIP session capabilities object.
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* \since 18.0.0
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*
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* \retval An ao2 allocated SIP session capabilities object, or NULL on error
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*/
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struct ast_sip_session_caps *ast_sip_session_caps_alloc(void);
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/*!
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* \brief Set the incoming call offer capabilities for a session.
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* \since 18.0.0
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*
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* This will replace any capabilities already present.
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*
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* \param caps A session's capabilities object
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* \param cap The capabilities to set it to
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*/
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void ast_sip_session_set_incoming_call_offer_cap(struct ast_sip_session_caps *caps,
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struct ast_format_cap *cap);
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/*!
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* \brief Get the incoming call offer capabilities.
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* \since 18.0.0
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*
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* \note Returned objects reference is not incremented.
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*
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* \param caps A session's capabilities object
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*
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* \retval An incoming call offer capabilities object
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*/
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const struct ast_format_cap *ast_sip_session_get_incoming_call_offer_cap(
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const struct ast_sip_session_caps *caps);
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/*!
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* \brief Make the incoming call offer capabilities for a session.
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* \since 18.0.0
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*
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* Creates and sets a list of joint capabilities between the given remote
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* capabilities, and pre-configured ones. The resulting joint list is then
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* stored, and 'owned' (reference held) by the session.
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*
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* If the incoming capabilities have been set elsewhere, this will not replace
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* those. It will however, return a pointer to the current set.
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*
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* \note Returned object's reference is not incremented.
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*
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* \param session The session
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* \param session_media An associated media session
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* \param remote Capabilities of a device
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*
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* \retval A pointer to the incoming call offer capabilities
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*/
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const struct ast_format_cap *ast_sip_session_join_incoming_call_offer_cap(
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const struct ast_sip_session *session, const struct ast_sip_session_media *session_media,
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const struct ast_format_cap *remote);
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#endif /* RES_PJSIP_SESSION_CAPS_H */
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@ -0,0 +1,162 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2020, Sangoma Technologies Corporation
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*
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* Kevin Harwell <kharwell@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#include "asterisk.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/channel.h"
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#include "asterisk/format.h"
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#include "asterisk/format_cap.h"
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#include "asterisk/logger.h"
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#include "asterisk/sorcery.h"
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#include <pjsip_ua.h>
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#include "asterisk/res_pjsip.h"
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#include "asterisk/res_pjsip_session.h"
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#include "asterisk/res_pjsip_session_caps.h"
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struct ast_sip_session_caps {
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struct ast_format_cap *incoming_call_offer_cap;
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};
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static void log_caps(int level, const char *file, int line, const char *function,
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const char *msg, const struct ast_sip_session *session,
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const struct ast_sip_session_media *session_media, const struct ast_format_cap *local,
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const struct ast_format_cap *remote, const struct ast_format_cap *joint)
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{
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struct ast_str *s1;
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struct ast_str *s2;
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struct ast_str *s3;
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if (level == __LOG_DEBUG && !DEBUG_ATLEAST(3)) {
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return;
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}
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s1 = local ? ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN) : NULL;
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s2 = remote ? ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN) : NULL;
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s3 = joint ? ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN) : NULL;
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ast_log(level, file, line, function, "'%s' %s '%s' capabilities -%s%s%s%s%s%s\n",
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session->channel ? ast_channel_name(session->channel) :
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ast_sorcery_object_get_id(session->endpoint),
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msg ? msg : "-", ast_codec_media_type2str(session_media->type),
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s1 ? " local: " : "", s1 ? ast_format_cap_get_names(local, &s1) : "",
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s2 ? " remote: " : "", s2 ? ast_format_cap_get_names(remote, &s2) : "",
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s3 ? " joint: " : "", s3 ? ast_format_cap_get_names(joint, &s3) : "");
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}
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static void sip_session_caps_destroy(void *obj)
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{
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struct ast_sip_session_caps *caps = obj;
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ao2_cleanup(caps->incoming_call_offer_cap);
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}
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struct ast_sip_session_caps *ast_sip_session_caps_alloc(void)
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{
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return ao2_alloc_options(sizeof(struct ast_sip_session_caps),
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sip_session_caps_destroy, AO2_ALLOC_OPT_LOCK_NOLOCK);
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}
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void ast_sip_session_set_incoming_call_offer_cap(struct ast_sip_session_caps *caps,
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struct ast_format_cap *cap)
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{
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ao2_cleanup(caps->incoming_call_offer_cap);
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caps->incoming_call_offer_cap = ao2_bump(cap);
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}
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const struct ast_format_cap *ast_sip_session_get_incoming_call_offer_cap(
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const struct ast_sip_session_caps *caps)
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{
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return caps->incoming_call_offer_cap;
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}
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const struct ast_format_cap *ast_sip_session_join_incoming_call_offer_cap(
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const struct ast_sip_session *session, const struct ast_sip_session_media *session_media,
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const struct ast_format_cap *remote)
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{
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enum ast_sip_call_codec_pref pref;
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struct ast_format_cap *joint;
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struct ast_format_cap *local;
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joint = session_media->caps->incoming_call_offer_cap;
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if (joint) {
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/*
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* If the incoming call offer capabilities have been set elsewhere, e.g. dialplan
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* then those take precedence.
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*/
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return joint;
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}
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joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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local = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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if (!joint || !local) {
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ast_log(LOG_ERROR, "Failed to allocate %s incoming call offer capabilities\n",
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ast_codec_media_type2str(session_media->type));
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ao2_cleanup(joint);
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ao2_cleanup(local);
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return NULL;
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}
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pref = session->endpoint->media.incoming_call_offer_pref;
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ast_format_cap_append_from_cap(local, session->endpoint->media.codecs,
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session_media->type);
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if (pref < AST_SIP_CALL_CODEC_PREF_REMOTE) {
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ast_format_cap_get_compatible(local, remote, joint); /* Prefer local */
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} else {
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ast_format_cap_get_compatible(remote, local, joint); /* Prefer remote */
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}
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if (ast_format_cap_empty(joint)) {
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log_caps(LOG_NOTICE, "No joint incoming", session, session_media, local, remote, NULL);
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ao2_ref(joint, -1);
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ao2_ref(local, -1);
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return NULL;
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}
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if (pref == AST_SIP_CALL_CODEC_PREF_LOCAL_SINGLE ||
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pref == AST_SIP_CALL_CODEC_PREF_REMOTE_SINGLE ||
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session->endpoint->preferred_codec_only) {
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/*
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* Save the most preferred one. Session capabilities are per stream and
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* a stream only carries a single media type, so no reason to worry with
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* the type here (i.e different or multiple types)
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*/
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struct ast_format *single = ast_format_cap_get_format(joint, 0);
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/* Remove all formats */
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ast_format_cap_remove_by_type(joint, AST_MEDIA_TYPE_UNKNOWN);
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/* Put the most preferred one back */
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ast_format_cap_append(joint, single, 0);
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ao2_ref(single, -1);
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}
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log_caps(LOG_DEBUG, "Joint incoming", session, session_media, local, remote, joint);
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ao2_ref(local, -1);
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ast_sip_session_set_incoming_call_offer_cap(session_media->caps, joint);
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return joint;
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}
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