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
changes/42/13842/3
Kevin Harwell 5 years ago
parent 87fda066ea
commit 06dada3f01

@ -798,6 +798,16 @@
; "0" or not enabled)
;contact_user= ; On outgoing requests, force the user portion of the Contact
; header to this value (default: "")
;incoming_call_offer_pref= ; Sets the preferred codecs, and order to use between
; those received in the offer, and those set in this
; configuration's allow line. Valid values include:
;
; local - prefer and order by configuration (default).
; local_single - prefer and order by configuration,
; but only choose 'top' most codec
; remote - prefer and order by incoming sdp.
; remote_single - prefer and order by incoming sdp,
; but only choose 'top' most codec
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.

@ -0,0 +1,53 @@
Subject: res_pjsip
Subject: res_pjsip_session
Master-Only: True
A new option, incoming_call_offer_pref, was added to res_pjsip endpoints that
specifies the preferred order of codecs to use between those received in the
offer, and those set in the configuration.
Valid values include:
local - prefer and order by configuration (default).
local_single - prefer and order by configuration, but only choose 'top'
most codec
remote - prefer and order by incoming sdp.
remote_single - prefer and order by incoming sdp, but only choose 'top' most
most codec
Example A:
[alice]
type=endpoint
incoming_call_offer_pref=local
allow=!all,opus,alaw,ulaw
Alice's incoming sdp=g722,ulaw,alaw
RESULT: alaw,ulaw
Example B:
[alice]
type=endpoint
incoming_call_offer_pref=local_single
allow=!all,opus,alaw,ulaw
Alice's incoming sdp=g722,ulaw,alaw
RESULT: alaw
Example C:
[alice]
type=endpoint
incoming_call_offer_pref=remote
allow=!all,opus,alaw,ulaw
Alice's incoming sdp=g722,ulaw,alaw
RESULT: ulaw,alaw
Example D:
[alice]
type=endpoint
incoming_call_offer_pref=remote_single
allow=!all,opus,alaw,ulaw
Alice's incoming sdp=g722,ulaw,alaw
RESULT: ulaw

@ -509,6 +509,24 @@ enum ast_sip_session_redirect {
AST_SIP_REDIRECT_URI_PJSIP,
};
/*!
* \brief Incoming/Outgoing call offer/answer joint codec preference.
*/
enum ast_sip_call_codec_pref {
/*! Prefer, and order by local values */
AST_SIP_CALL_CODEC_PREF_LOCAL,
/*! Prefer, and order by local values (intersection) */
AST_SIP_CALL_CODEC_PREF_LOCAL_LIMIT,
/*! Prefer, and order by local values (top/first only) */
AST_SIP_CALL_CODEC_PREF_LOCAL_SINGLE,
/*! Prefer, and order by remote values */
AST_SIP_CALL_CODEC_PREF_REMOTE,
/*! Prefer, and order by remote values (intersection) */
AST_SIP_CALL_CODEC_PREF_REMOTE_LIMIT,
/*! Prefer, and order by remote values (top/first only) */
AST_SIP_CALL_CODEC_PREF_REMOTE_SINGLE,
};
/*!
* \brief Session timers options
*/
@ -750,6 +768,8 @@ struct ast_sip_endpoint_media_configuration {
unsigned int bundle;
/*! Enable webrtc settings and defaults */
unsigned int webrtc;
/*! Codec preference for an incoming offer */
enum ast_sip_call_codec_pref incoming_call_offer_pref;
};
/*!

@ -59,6 +59,7 @@ enum ast_sip_session_t38state {
struct ast_sip_session_sdp_handler;
struct ast_sip_session;
struct ast_sip_session_caps;
struct ast_sip_session_media;
typedef struct ast_frame *(*ast_sip_session_media_read_cb)(struct ast_sip_session *session, struct ast_sip_session_media *session_media);
@ -79,6 +80,8 @@ struct ast_sip_session_media {
struct ast_sip_session_sdp_handler *handler;
/*! \brief Holds SRTP information */
struct ast_sdp_srtp *srtp;
/*! \brief Media format capabilities */
struct ast_sip_session_caps *caps;
/*! \brief What type of encryption is in use on this stream */
enum ast_sip_session_media_encryption encryption;
/*! \brief The media transport in use for this stream */

@ -0,0 +1,82 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2020, Sangoma Technologies Corporation
*
* Kevin Harwell <kharwell@sangoma.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.
*/
#ifndef RES_PJSIP_SESSION_CAPS_H
#define RES_PJSIP_SESSION_CAPS_H
struct ast_format_cap;
struct ast_sip_session;
struct ast_sip_session_media;
struct ast_sip_session_caps;
/*!
* \brief Allocate a SIP session capabilities object.
* \since 18.0.0
*
* \retval An ao2 allocated SIP session capabilities object, or NULL on error
*/
struct ast_sip_session_caps *ast_sip_session_caps_alloc(void);
/*!
* \brief Set the incoming call offer capabilities for a session.
* \since 18.0.0
*
* This will replace any capabilities already present.
*
* \param caps A session's capabilities object
* \param cap The capabilities to set it to
*/
void ast_sip_session_set_incoming_call_offer_cap(struct ast_sip_session_caps *caps,
struct ast_format_cap *cap);
/*!
* \brief Get the incoming call offer capabilities.
* \since 18.0.0
*
* \note Returned objects reference is not incremented.
*
* \param caps A session's capabilities object
*
* \retval An incoming call offer capabilities object
*/
const struct ast_format_cap *ast_sip_session_get_incoming_call_offer_cap(
const struct ast_sip_session_caps *caps);
/*!
* \brief Make the incoming call offer capabilities for a session.
* \since 18.0.0
*
* Creates and sets a list of joint capabilities between the given remote
* capabilities, and pre-configured ones. The resulting joint list is then
* stored, and 'owned' (reference held) by the session.
*
* If the incoming capabilities have been set elsewhere, this will not replace
* those. It will however, return a pointer to the current set.
*
* \note Returned object's reference is not incremented.
*
* \param session The session
* \param session_media An associated media session
* \param remote Capabilities of a device
*
* \retval A pointer to the incoming call offer capabilities
*/
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);
#endif /* RES_PJSIP_SESSION_CAPS_H */

@ -66,6 +66,7 @@ $(call MOD_ADD_C,res_stasis,$(wildcard stasis/*.c))
$(call MOD_ADD_C,res_snmp,snmp/agent.c)
$(call MOD_ADD_C,res_parking,$(wildcard parking/*.c))
$(call MOD_ADD_C,res_pjsip,$(wildcard res_pjsip/*.c))
$(call MOD_ADD_C,res_pjsip_session,$(wildcard res_pjsip_session/*.c))
$(call MOD_ADD_C,res_prometheus,$(wildcard prometheus/*.c))
$(call MOD_ADD_C,res_ari,ari/cli.c ari/config.c ari/ari_websockets.c)
$(call MOD_ADD_C,res_ari_model,ari/ari_model_validators.c)

@ -925,6 +925,27 @@
<configOption name="preferred_codec_only" default="no">
<synopsis>Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.</synopsis>
</configOption>
<configOption name="incoming_call_offer_pref" default="local">
<synopsis>After receiving an incoming offer create a list of preferred codecs between
those received in the SDP offer, and those specified in endpoint configuration.</synopsis>
<description>
<note><para>This list will consist of only those codecs found in both.</para></note>
<enumlist>
<enum name="local"><para>
Order by the endpoint configuration allow line (default)
</para></enum>
<enum name="local_single"><para>
Order by the endpoint configuration allow line, but the list will only contain the first, or 'top' item
</para></enum>
<enum name="remote"><para>
Order by what is received in the SDP offer
</para></enum>
<enum name="remote_single"><para>
Order by what is received in the SDP offer, but the list will only contain the first, or 'top' item
</para></enum>
</enumlist>
</description>
</configOption>
<configOption name="rtp_keepalive">
<synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
<description><para>

@ -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");

@ -56,6 +56,7 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/res_pjsip_session_caps.h"
/*! \brief Scheduler for RTCP purposes */
static struct ast_sched_context *sched;
@ -373,6 +374,81 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
}
}
static int apply_cap_to_bundled(struct ast_sip_session_media *session_media,
struct ast_sip_session_media *session_media_transport,
struct ast_stream *asterisk_stream, const struct ast_format_cap *joint)
{
if (!joint) {
return -1;
}
ast_stream_set_formats(asterisk_stream, (struct ast_format_cap *)joint);
/* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
if (session_media_transport != session_media && session_media->bundled) {
int index;
for (index = 0; index < ast_format_cap_count(joint); ++index) {
struct ast_format *format = ast_format_cap_get_format(joint, index);
int rtp_code;
/* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
* things as the format is guaranteed to have a payload already.
*/
rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
ao2_ref(format, -1);
}
}
return 0;
}
static const struct ast_format_cap *set_incoming_call_offer_cap(
struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *stream)
{
const struct ast_format_cap *incoming_call_offer_cap;
struct ast_format_cap *remote;
struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
int fmts = 0;
remote = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!remote) {
ast_log(LOG_ERROR, "Failed to allocate %s incoming remote capabilities\n",
ast_codec_media_type2str(session_media->type));
return NULL;
}
/* Get the peer's capabilities*/
get_codecs(session, stream, &codecs, session_media);
ast_rtp_codecs_payload_formats(&codecs, remote, &fmts);
incoming_call_offer_cap = ast_sip_session_join_incoming_call_offer_cap(
session, session_media, remote);
ao2_ref(remote, -1);
if (!incoming_call_offer_cap) {
ast_rtp_codecs_payloads_destroy(&codecs);
return NULL;
}
/*
* Setup rx payload type mapping to prefer the mapping
* from the peer that the RFC says we SHOULD use.
*/
ast_rtp_codecs_payloads_xover(&codecs, &codecs, NULL);
ast_rtp_codecs_payloads_copy(&codecs,
ast_rtp_instance_get_codecs(session_media->rtp), session_media->rtp);
ast_rtp_codecs_payloads_destroy(&codecs);
return incoming_call_offer_cap;
}
static int set_caps(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
struct ast_sip_session_media *session_media_transport,
@ -432,25 +508,7 @@ static int set_caps(struct ast_sip_session *session,
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
ast_stream_set_formats(asterisk_stream, joint);
/* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
if (session_media_transport != session_media && session_media->bundled) {
int index;
for (index = 0; index < ast_format_cap_count(joint); ++index) {
struct ast_format *format = ast_format_cap_get_format(joint, index);
int rtp_code;
/* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
* things as the format is guaranteed to have a payload already.
*/
rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
ao2_ref(format, -1);
}
}
apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
if (session->channel && ast_sip_session_is_pending_stream_default(session, asterisk_stream)) {
ast_channel_lock(session->channel);
@ -1420,7 +1478,8 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
session_media->remotely_held_changed = 1;
}
if (set_caps(session, session_media, session_media_transport, stream, 1, asterisk_stream)) {
if (apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream,
set_incoming_call_offer_cap(session, session_media, stream))) {
return 0;
}

@ -30,6 +30,7 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/res_pjsip_session_caps.h"
#include "asterisk/callerid.h"
#include "asterisk/datastore.h"
#include "asterisk/module.h"
@ -466,6 +467,8 @@ static void session_media_dtor(void *obj)
ast_free(session_media->mid);
ast_free(session_media->remote_mslabel);
ao2_cleanup(session_media->caps);
}
struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_session *session,
@ -524,6 +527,12 @@ struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_ses
} else {
session_media->bundle_group = -1;
}
session_media->caps = ast_sip_session_caps_alloc();
if (!session_media->caps) {
ao2_ref(session_media, -1);
return NULL;
}
}
if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {

@ -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;
}
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