[svn-commits] file: trunk r369769 - in /trunk: ./ channels/ configs/ include/asterisk/ res/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sat Jul 7 12:07:00 CDT 2012
Author: file
Date: Sat Jul 7 12:06:51 2012
New Revision: 369769
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369769
Log:
Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif.
This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.
You may ask yourself though where the name motif comes from... and I would say to you... music!
motif: a perceivable or salient recurring fragment or succession of notes
Sorta like a jingle!
Review: https://reviewboard.asterisk.org/r/1917/
Added:
trunk/channels/chan_motif.c (with props)
trunk/configs/motif.conf.sample (with props)
trunk/res/res_xmpp.exports.in (with props)
Modified:
trunk/CHANGES
trunk/UPGRADE.txt
trunk/channels/chan_gtalk.c
trunk/include/asterisk/xmpp.h
trunk/res/res_jabber.c
trunk/res/res_xmpp.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=369769&r1=369768&r2=369769
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Sat Jul 7 12:06:51 2012
@@ -294,6 +294,15 @@
-----------
* Direct media functionality has been added.
Options in config are: directmedia (directrtp) and directrtpsetup (earlydirect)
+
+chan_motif
+----------
+ * A new channel driver named chan_motif has been added which provides support for
+ Google Talk and Jingle in a single channel driver. This new channel driver includes
+ support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
+ hold, unhold, and ringing notification. It is also compliant with the current Jingle
+ specification, current Google Jingle specification, and the original Google Talk
+ protocol.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
Modified: trunk/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE.txt?view=diff&rev=369769&r1=369768&r2=369769
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Sat Jul 7 12:06:51 2012
@@ -88,6 +88,14 @@
- Answered outgoing calls no longer get cut off when the next step is started.
You now have until the last step times out to decide if you want to accept
the call or not before being disconnected.
+
+chan_gtalk:
+ - chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
+ that users switch to using it as it is a core supported module.
+
+chan_jingle:
+ - chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
+ that users switch to using it as it is a core supported module.
SIP
===
Modified: trunk/channels/chan_gtalk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_gtalk.c?view=diff&rev=369769&r1=369768&r2=369769
==============================================================================
--- trunk/channels/chan_gtalk.c (original)
+++ trunk/channels/chan_gtalk.c Sat Jul 7 12:06:51 2012
@@ -32,6 +32,7 @@
*/
/*** MODULEINFO
+ <defaultenabled>no</defaultenabled>
<depend>iksemel</depend>
<depend>res_jabber</depend>
<use type="external">openssl</use>
Added: trunk/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_motif.c?view=auto&rev=369769
==============================================================================
--- trunk/channels/chan_motif.c (added)
+++ trunk/channels/chan_motif.c Sat Jul 7 12:06:51 2012
@@ -1,0 +1,2515 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at 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.
+ */
+
+/*! \file
+ *
+ * \author Joshua Colp <jcolp at digium.com>
+ *
+ * \brief Motif Jingle Channel Driver
+ *
+ * \extref Iksemel http://iksemel.jabberstudio.org/
+ *
+ * \ingroup channel_drivers
+ */
+
+/*** MODULEINFO
+ <depend>iksemel</depend>
+ <depend>res_jabber</depend>
+ <use type="external">openssl</use>
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <sys/socket.h>
+#include <fcntl.h>
+#include <netdb.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <sys/signal.h>
+#include <iksemel.h>
+#include <pthread.h>
+
+#include "asterisk/lock.h"
+#include "asterisk/channel.h"
+#include "asterisk/config_options.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/sched.h"
+#include "asterisk/io.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/acl.h"
+#include "asterisk/callerid.h"
+#include "asterisk/file.h"
+#include "asterisk/cli.h"
+#include "asterisk/app.h"
+#include "asterisk/musiconhold.h"
+#include "asterisk/manager.h"
+#include "asterisk/stringfields.h"
+#include "asterisk/utils.h"
+#include "asterisk/causes.h"
+#include "asterisk/astobj.h"
+#include "asterisk/abstract_jb.h"
+#include "asterisk/xmpp.h"
+
+/*! \brief Default maximum number of ICE candidates we will offer */
+#define DEFAULT_MAX_ICE_CANDIDATES "10"
+
+/*! \brief Default maximum number of payloads we will offer */
+#define DEFAULT_MAX_PAYLOADS "30"
+
+/*! \brief Number of buckets for endpoints */
+#define ENDPOINT_BUCKETS 37
+
+/*! \brief Number of buckets for sessions, on a per-endpoint basis */
+#define SESSION_BUCKETS 37
+
+/*! \brief Namespace for Jingle itself */
+#define JINGLE_NS "urn:xmpp:jingle:1"
+
+/*! \brief Namespace for Jingle RTP sessions */
+#define JINGLE_RTP_NS "urn:xmpp:jingle:apps:rtp:1"
+
+/*! \brief Namespace for Jingle RTP info */
+#define JINGLE_RTP_INFO_NS "urn:xmpp:jingle:apps:rtp:info:1"
+
+/*! \brief Namespace for Jingle ICE-UDP */
+#define JINGLE_ICE_UDP_NS "urn:xmpp:jingle:transports:ice-udp:1"
+
+/*! \brief Namespace for Google Talk ICE-UDP */
+#define GOOGLE_TRANSPORT_NS "http://www.google.com/transport/p2p"
+
+/*! \brief Namespace for Google Talk Raw UDP */
+#define GOOGLE_TRANSPORT_RAW_NS "http://www.google.com/transport/raw-udp"
+
+/*! \brief Namespace for Google Session */
+#define GOOGLE_SESSION_NS "http://www.google.com/session"
+
+/*! \brief Namespace for Google Phone description */
+#define GOOGLE_PHONE_NS "http://www.google.com/session/phone"
+
+/*! \brief Namespace for Google Video description */
+#define GOOGLE_VIDEO_NS "http://www.google.com/session/video"
+
+/*! \brief Namespace for XMPP stanzas */
+#define XMPP_STANZAS_NS "urn:ietf:params:xml:ns:xmpp-stanzas"
+
+/*! \brief The various transport methods supported, from highest priority to lowest priority when doing fallback */
+enum jingle_transport {
+ JINGLE_TRANSPORT_ICE_UDP = 3, /*!< XEP-0176 */
+ JINGLE_TRANSPORT_GOOGLE_V2 = 2, /*!< https://developers.google.com/talk/call_signaling */
+ JINGLE_TRANSPORT_GOOGLE_V1 = 1, /*!< Undocumented initial Google specification */
+ JINGLE_TRANSPORT_NONE = 0, /*!< No transport specified */
+};
+
+/*! \brief Endpoint state information */
+struct jingle_endpoint_state {
+ struct ao2_container *sessions; /*!< Active sessions to or from the endpoint */
+};
+
+/*! \brief Endpoint which contains configuration information and active sessions */
+struct jingle_endpoint {
+ AST_DECLARE_STRING_FIELDS(
+ AST_STRING_FIELD(name); /*!< Name of the endpoint */
+ AST_STRING_FIELD(context); /*!< Context to place incoming calls into */
+ AST_STRING_FIELD(accountcode); /*!< Account code */
+ AST_STRING_FIELD(language); /*!< Default language for prompts */
+ AST_STRING_FIELD(musicclass); /*!< Configured music on hold class */
+ AST_STRING_FIELD(parkinglot); /*!< Configured parking lot */
+ );
+ struct ast_xmpp_client *connection; /*!< Connection to use for traffic */
+ iksrule *rule; /*!< Active matching rule */
+ unsigned int maxicecandidates; /*!< Maximum number of ICE candidates we will offer */
+ unsigned int maxpayloads; /*!< Maximum number of payloads we will offer */
+ struct ast_codec_pref prefs; /*!< Codec preferences */
+ struct ast_format_cap *cap; /*!< Formats to use */
+ ast_group_t callgroup; /*!< Call group */
+ ast_group_t pickupgroup; /*!< Pickup group */
+ enum jingle_transport transport; /*!< Default transport to use on outgoing sessions */
+ struct jingle_endpoint_state *state; /*!< Endpoint state information */
+};
+
+/*! \brief Session which contains information about an active session */
+struct jingle_session {
+ AST_DECLARE_STRING_FIELDS(
+ AST_STRING_FIELD(sid); /*!< Session identifier */
+ AST_STRING_FIELD(audio_name); /*!< Name of the audio content */
+ AST_STRING_FIELD(video_name); /*!< Name of the video content */
+ );
+ struct jingle_endpoint_state *state; /*!< Endpoint we are associated with */
+ struct ast_xmpp_client *connection; /*!< Connection to use for traffic */
+ enum jingle_transport transport; /*!< Transport type to use for this session */
+ unsigned int maxicecandidates; /*!< Maximum number of ICE candidates we will offer */
+ unsigned int maxpayloads; /*!< Maximum number of payloads we will offer */
+ char remote_original[XMPP_MAX_JIDLEN];/*!< Identifier of the original remote party (remote may have changed due to redirect) */
+ char remote[XMPP_MAX_JIDLEN]; /*!< Identifier of the remote party */
+ iksrule *rule; /*!< Session matching rule */
+ struct ast_codec_pref prefs; /*!< Codec preferences */
+ struct ast_channel *owner; /*!< Master Channel */
+ struct ast_rtp_instance *rtp; /*!< RTP audio session */
+ struct ast_rtp_instance *vrtp; /*!< RTP video session */
+ struct ast_format_cap *cap; /*!< Local codec capabilities */
+ struct ast_format_cap *jointcap; /*!< Joint codec capabilities */
+ struct ast_format_cap *peercap; /*!< Peer codec capabilities */
+ unsigned int outgoing:1; /*!< Whether this is an outgoing leg or not */
+ unsigned int gone:1; /*!< In the eyes of Jingle this session is already gone */
+};
+
+static const char desc[] = "Motif Jingle Channel";
+static const char channel_type[] = "Motif";
+
+struct jingle_config {
+ struct ao2_container *endpoints; /*!< Configured endpoints */
+};
+
+static AO2_GLOBAL_OBJ_STATIC(globals);
+
+static struct ast_sched_context *sched; /*!< Scheduling context for RTCP */
+
+/* \brief Asterisk core interaction functions */
+static struct ast_channel *jingle_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
+static int jingle_sendtext(struct ast_channel *ast, const char *text);
+static int jingle_digit_begin(struct ast_channel *ast, char digit);
+static int jingle_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
+static int jingle_call(struct ast_channel *ast, const char *dest, int timeout);
+static int jingle_hangup(struct ast_channel *ast);
+static int jingle_answer(struct ast_channel *ast);
+static struct ast_frame *jingle_read(struct ast_channel *ast);
+static int jingle_write(struct ast_channel *ast, struct ast_frame *f);
+static int jingle_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
+static int jingle_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
+static struct jingle_session *jingle_alloc(struct jingle_endpoint *endpoint, const char *from, const char *sid);
+
+/*! \brief Action handlers */
+static void jingle_action_session_initiate(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
+static void jingle_action_transport_info(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
+static void jingle_action_session_accept(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
+static void jingle_action_session_info(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
+static void jingle_action_session_terminate(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
+
+/*! \brief PBX interface structure for channel registration */
+static struct ast_channel_tech jingle_tech = {
+ .type = "Motif",
+ .description = "Motif Jingle Channel Driver",
+ .requester = jingle_request,
+ .send_text = jingle_sendtext,
+ .send_digit_begin = jingle_digit_begin,
+ .send_digit_end = jingle_digit_end,
+ .bridge = ast_rtp_instance_bridge,
+ .call = jingle_call,
+ .hangup = jingle_hangup,
+ .answer = jingle_answer,
+ .read = jingle_read,
+ .write = jingle_write,
+ .write_video = jingle_write,
+ .exception = jingle_read,
+ .indicate = jingle_indicate,
+ .fixup = jingle_fixup,
+ .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
+};
+
+/*! \brief Defined handlers for different Jingle actions */
+static const struct jingle_action_handler {
+ const char *action;
+ void (*handler)(struct jingle_endpoint *endpoint, struct jingle_session *session, ikspak *pak);
+} jingle_action_handlers[] = {
+ /* Jingle actions */
+ { "session-initiate", jingle_action_session_initiate, },
+ { "transport-info", jingle_action_transport_info, },
+ { "session-accept", jingle_action_session_accept, },
+ { "session-info", jingle_action_session_info, },
+ { "session-terminate", jingle_action_session_terminate, },
+ /* Google-V1 actions */
+ { "initiate", jingle_action_session_initiate, },
+ { "candidates", jingle_action_transport_info, },
+ { "accept", jingle_action_session_accept, },
+ { "terminate", jingle_action_session_terminate, },
+ { "reject", jingle_action_session_terminate, },
+};
+
+/*! \brief Reason text <-> cause code mapping */
+static const struct jingle_reason_mapping {
+ const char *reason;
+ int cause;
+} jingle_reason_mappings[] = {
+ { "busy", AST_CAUSE_BUSY, },
+ { "cancel", AST_CAUSE_CALL_REJECTED, },
+ { "connectivity-error", AST_CAUSE_INTERWORKING, },
+ { "decline", AST_CAUSE_CALL_REJECTED, },
+ { "expired", AST_CAUSE_NO_USER_RESPONSE, },
+ { "failed-transport", AST_CAUSE_PROTOCOL_ERROR, },
+ { "failed-application", AST_CAUSE_SWITCH_CONGESTION, },
+ { "general-error", AST_CAUSE_CONGESTION, },
+ { "gone", AST_CAUSE_NORMAL_CLEARING, },
+ { "incompatible-parameters", AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, },
+ { "media-error", AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, },
+ { "security-error", AST_CAUSE_PROTOCOL_ERROR, },
+ { "success", AST_CAUSE_NORMAL_CLEARING, },
+ { "timeout", AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, },
+ { "unsupported-applications", AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, },
+ { "unsupported-transports", AST_CAUSE_FACILITY_NOT_IMPLEMENTED, },
+};
+
+/*! \brief Hashing function for Jingle sessions */
+static int jingle_session_hash(const void *obj, const int flags)
+{
+ const struct jingle_session *session = obj;
+ const char *sid = obj;
+
+ return ast_str_hash(flags & OBJ_KEY ? sid : session->sid);
+}
+
+/*! \brief Comparator function for Jingle sessions */
+static int jingle_session_cmp(void *obj, void *arg, int flags)
+{
+ struct jingle_session *session1 = obj, *session2 = arg;
+ const char *sid = arg;
+
+ return !strcmp(session1->sid, flags & OBJ_KEY ? sid : session2->sid) ? CMP_MATCH | CMP_STOP : 0;
+}
+
+/*! \brief Destructor for Jingle endpoint state */
+static void jingle_endpoint_state_destructor(void *obj)
+{
+ struct jingle_endpoint_state *state = obj;
+
+ ao2_ref(state->sessions, -1);
+}
+
+/*! \brief Destructor for Jingle endpoints */
+static void jingle_endpoint_destructor(void *obj)
+{
+ struct jingle_endpoint *endpoint = obj;
+
+ if (endpoint->rule) {
+ iks_filter_remove_rule(endpoint->connection->filter, endpoint->rule);
+ }
+
+ if (endpoint->connection) {
+ ast_xmpp_client_unref(endpoint->connection);
+ }
+
+ ast_format_cap_destroy(endpoint->cap);
+
+ ao2_ref(endpoint->state, -1);
+
+ ast_string_field_free_memory(endpoint);
+}
+
+/*! \brief Find function for Jingle endpoints */
+static void *jingle_endpoint_find(struct ao2_container *tmp_container, const char *category)
+{
+ return ao2_find(tmp_container, category, OBJ_KEY);
+}
+
+/*! \brief Allocator function for Jingle endpoint state */
+static struct jingle_endpoint_state *jingle_endpoint_state_create(void)
+{
+ struct jingle_endpoint_state *state;
+
+ if (!(state = ao2_alloc(sizeof(*state), jingle_endpoint_state_destructor))) {
+ return NULL;
+ }
+
+ if (!(state->sessions = ao2_container_alloc(SESSION_BUCKETS, jingle_session_hash, jingle_session_cmp))) {
+ ao2_ref(state, -1);
+ return NULL;
+ }
+
+ return state;
+}
+
+/*! \brief State find/create function */
+static struct jingle_endpoint_state *jingle_endpoint_state_find_or_create(const char *category)
+{
+ RAII_VAR(struct jingle_config *, cfg, ao2_global_obj_ref(globals), ao2_cleanup);
+ RAII_VAR(struct jingle_endpoint *, endpoint, NULL, ao2_cleanup);
+
+ if (!cfg || !cfg->endpoints || !(endpoint = jingle_endpoint_find(cfg->endpoints, category))) {
+ return jingle_endpoint_state_create();
+ }
+
+ ao2_ref(endpoint->state, +1);
+ return endpoint->state;
+}
+
+/*! \brief Allocator function for Jingle endpoints */
+static void *jingle_endpoint_alloc(const char *cat)
+{
+ struct jingle_endpoint *endpoint;
+
+ if (!(endpoint = ao2_alloc(sizeof(*endpoint), jingle_endpoint_destructor))) {
+ return NULL;
+ }
+
+ if (ast_string_field_init(endpoint, 512)) {
+ ao2_ref(endpoint, -1);
+ return NULL;
+ }
+
+ if (!(endpoint->state = jingle_endpoint_state_find_or_create(cat))) {
+ ao2_ref(endpoint, -1);
+ return NULL;
+ }
+
+ ast_string_field_set(endpoint, name, cat);
+
+ endpoint->cap = ast_format_cap_alloc_nolock();
+ endpoint->transport = JINGLE_TRANSPORT_ICE_UDP;
+
+ return endpoint;
+}
+
+/*! \brief Hashing function for Jingle endpoints */
+static int jingle_endpoint_hash(const void *obj, const int flags)
+{
+ const struct jingle_endpoint *endpoint = obj;
+ const char *name = obj;
+
+ return ast_str_hash(flags & OBJ_KEY ? name : endpoint->name);
+}
+
+/*! \brief Comparator function for Jingle endpoints */
+static int jingle_endpoint_cmp(void *obj, void *arg, int flags)
+{
+ struct jingle_endpoint *endpoint1 = obj, *endpoint2 = arg;
+ const char *name = arg;
+
+ return !strcmp(endpoint1->name, flags & OBJ_KEY ? name : endpoint2->name) ? CMP_MATCH | CMP_STOP : 0;
+}
+
+static struct aco_type endpoint_option = {
+ .type = ACO_ITEM,
+ .category_match = ACO_BLACKLIST,
+ .category = "^general$",
+ .item_alloc = jingle_endpoint_alloc,
+ .item_find = jingle_endpoint_find,
+ .item_offset = offsetof(struct jingle_config, endpoints),
+};
+
+struct aco_type *endpoint_options[] = ACO_TYPES(&endpoint_option);
+
+struct aco_file jingle_conf = {
+ .filename = "motif.conf",
+ .types = ACO_TYPES(&endpoint_option),
+};
+
+/*! \brief Destructor for Jingle sessions */
+static void jingle_session_destructor(void *obj)
+{
+ struct jingle_session *session = obj;
+
+ if (session->rule) {
+ iks_filter_remove_rule(session->connection->filter, session->rule);
+ }
+
+ if (session->connection) {
+ ast_xmpp_client_unref(session->connection);
+ }
+
+ if (session->rtp) {
+ ast_rtp_instance_destroy(session->rtp);
+ }
+
+ if (session->vrtp) {
+ ast_rtp_instance_destroy(session->vrtp);
+ }
+
+ ast_format_cap_destroy(session->cap);
+ ast_format_cap_destroy(session->jointcap);
+ ast_format_cap_destroy(session->peercap);
+
+ ast_string_field_free_memory(session);
+}
+
+/*! \brief Destructor called when module configuration goes away */
+static void jingle_config_destructor(void *obj)
+{
+ struct jingle_config *cfg = obj;
+ ao2_cleanup(cfg->endpoints);
+}
+
+/*! \brief Allocator called when module configuration should appear */
+static void *jingle_config_alloc(void)
+{
+ struct jingle_config *cfg;
+
+ if (!(cfg = ao2_alloc(sizeof(*cfg), jingle_config_destructor))) {
+ return NULL;
+ }
+
+ if (!(cfg->endpoints = ao2_container_alloc(ENDPOINT_BUCKETS, jingle_endpoint_hash, jingle_endpoint_cmp))) {
+ ao2_ref(cfg, -1);
+ return NULL;
+ }
+
+ return cfg;
+}
+
+CONFIG_INFO_STANDARD(cfg_info, globals, jingle_config_alloc,
+ .files = ACO_FILES(&jingle_conf),
+ );
+
+/*! \brief Function called by RTP engine to get local RTP peer */
+static enum ast_rtp_glue_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
+{
+ struct jingle_session *session = ast_channel_tech_pvt(chan);
+ enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
+
+ if (!session->rtp) {
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
+
+ ao2_ref(session->rtp, +1);
+ *instance = session->rtp;
+
+ return res;
+}
+
+/*! \brief Function called by RTP engine to get peer capabilities */
+static void jingle_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
+{
+}
+
+/*! \brief Function called by RTP engine to change where the remote party should send media */
+static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active)
+{
+ return -1;
+}
+
+/*! \brief Local glue for interacting with the RTP engine core */
+static struct ast_rtp_glue jingle_rtp_glue = {
+ .type = "Motif",
+ .get_rtp_info = jingle_get_rtp_peer,
+ .get_codec = jingle_get_codec,
+ .update_peer = jingle_set_rtp_peer,
+};
+
+/*! \brief Internal helper function which enables video support on a sesson if possible */
+static void jingle_enable_video(struct jingle_session *session)
+{
+ struct ast_sockaddr tmp;
+ struct ast_rtp_engine_ice *ice;
+
+ /* If video is already present don't do anything */
+ if (session->vrtp) {
+ return;
+ }
+
+ /* If there are no configured video codecs do not turn video support on, it just won't work */
+ if (!ast_format_cap_has_type(session->cap, AST_FORMAT_TYPE_VIDEO)) {
+ return;
+ }
+
+ ast_sockaddr_parse(&tmp, "0.0.0.0", 0);
+
+ if (!(session->vrtp = ast_rtp_instance_new("asterisk", sched, &tmp, NULL))) {
+ return;
+ }
+
+ ast_rtp_instance_set_prop(session->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+
+ ast_channel_set_fd(session->owner, 2, ast_rtp_instance_fd(session->vrtp, 0));
+ ast_channel_set_fd(session->owner, 3, ast_rtp_instance_fd(session->vrtp, 1));
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->vrtp), session->vrtp, &session->prefs);
+
+ if (session->transport == JINGLE_TRANSPORT_GOOGLE_V2 && (ice = ast_rtp_instance_get_ice(session->vrtp))) {
+ ice->stop(session->vrtp);
+ }
+}
+
+/*! \brief Internal helper function used to allocate Jingle session on an endpoint */
+static struct jingle_session *jingle_alloc(struct jingle_endpoint *endpoint, const char *from, const char *sid)
+{
+ struct jingle_session *session;
+ struct ast_sockaddr tmp;
+
+ if (!(session = ao2_alloc(sizeof(*session), jingle_session_destructor))) {
+ return NULL;
+ }
+
+ if (ast_string_field_init(session, 512)) {
+ ao2_ref(session, -1);
+ return NULL;
+ }
+
+ if (!ast_strlen_zero(from)) {
+ ast_copy_string(session->remote_original, from, sizeof(session->remote_original));
+ ast_copy_string(session->remote, from, sizeof(session->remote));
+ }
+
+ if (ast_strlen_zero(sid)) {
+ ast_string_field_build(session, sid, "%08lx%08lx", ast_random(), ast_random());
+ session->outgoing = 1;
+ ast_string_field_set(session, audio_name, "audio");
+ ast_string_field_set(session, video_name, "video");
+ } else {
+ ast_string_field_set(session, sid, sid);
+ }
+
+ ao2_ref(endpoint->state, +1);
+ session->state = endpoint->state;
+ ao2_ref(endpoint->connection, +1);
+ session->connection = endpoint->connection;
+ session->transport = endpoint->transport;
+
+ if (!(session->cap = ast_format_cap_alloc_nolock()) ||
+ !(session->jointcap = ast_format_cap_alloc_nolock()) ||
+ !(session->peercap = ast_format_cap_alloc_nolock())) {
+ ao2_ref(session, -1);
+ return NULL;
+ }
+
+ ast_format_cap_copy(session->cap, endpoint->cap);
+
+ /* While we rely on res_jabber for communication we still need a temporary ast_sockaddr to tell the RTP engine
+ * that we want IPv4 */
+ ast_sockaddr_parse(&tmp, "0.0.0.0", 0);
+
+ /* Sessions always carry audio, but video is optional so don't enable it here */
+ if (!(session->rtp = ast_rtp_instance_new("asterisk", sched, &tmp, NULL))) {
+ ao2_ref(session, -1);
+ return NULL;
+ }
+ ast_rtp_instance_set_prop(session->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(session->rtp, AST_RTP_PROPERTY_DTMF, 1);
+
+ memcpy(&session->prefs, &endpoint->prefs, sizeof(session->prefs));
+
+ session->maxicecandidates = endpoint->maxicecandidates;
+ session->maxpayloads = endpoint->maxpayloads;
+
+ return session;
+}
+
+/*! \brief Function called to create a new Jingle Asterisk channel */
+static struct ast_channel *jingle_new(struct jingle_endpoint *endpoint, struct jingle_session *session, int state, const char *title, const char *linkedid, const char *cid_name)
+{
+ struct ast_channel *chan;
+ const char *str = S_OR(title, session->remote);
+ struct ast_format tmpfmt;
+
+ if (ast_format_cap_is_empty(session->cap)) {
+ return NULL;
+ }
+
+ if (!(chan = ast_channel_alloc(1, state, S_OR(title, ""), S_OR(cid_name, ""), "", "", "", linkedid, 0, "Motif/%s-%04lx", str, ast_random() & 0xffff))) {
+ return NULL;
+ }
+
+ ast_channel_tech_set(chan, &jingle_tech);
+ ast_channel_tech_pvt_set(chan, session);
+ session->owner = chan;
+
+ ast_format_cap_copy(ast_channel_nativeformats(chan), session->cap);
+ ast_codec_choose(&session->prefs, session->cap, 1, &tmpfmt);
+
+ if (session->rtp) {
+ struct ast_rtp_engine_ice *ice;
+
+ ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(session->rtp, 0));
+ ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(session->rtp, 1));
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session->rtp), session->rtp, &session->prefs);
+
+ if (((session->transport == JINGLE_TRANSPORT_GOOGLE_V2) ||
+ (session->transport == JINGLE_TRANSPORT_GOOGLE_V1)) &&
+ (ice = ast_rtp_instance_get_ice(session->rtp))) {
+ /* We stop built in ICE support because we need to fall back to old old old STUN support */
+ ice->stop(session->rtp);
+ }
+ }
+
+ if (state == AST_STATE_RING) {
+ ast_channel_rings_set(chan, 1);
+ }
+
+ ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
+
+ ast_best_codec(ast_channel_nativeformats(chan), &tmpfmt);
+ ast_format_copy(ast_channel_writeformat(chan), &tmpfmt);
+ ast_format_copy(ast_channel_rawwriteformat(chan), &tmpfmt);
+ ast_format_copy(ast_channel_readformat(chan), &tmpfmt);
+ ast_format_copy(ast_channel_rawreadformat(chan), &tmpfmt);
+
+ ao2_lock(endpoint);
+
+ ast_channel_callgroup_set(chan, endpoint->callgroup);
+ ast_channel_pickupgroup_set(chan, endpoint->pickupgroup);
+
+ if (!ast_strlen_zero(endpoint->accountcode)) {
+ ast_channel_accountcode_set(chan, endpoint->accountcode);
+ }
+
+ if (!ast_strlen_zero(endpoint->language)) {
+ ast_channel_language_set(chan, endpoint->language);
+ }
+
+ if (!ast_strlen_zero(endpoint->musicclass)) {
+ ast_channel_musicclass_set(chan, endpoint->musicclass);
+ }
+
+ ast_channel_context_set(chan, endpoint->context);
+ ast_channel_exten_set(chan, "s");
+ ast_channel_priority_set(chan, 1);
+
+ ao2_unlock(endpoint);
+
+ return chan;
+}
+
+/*! \brief Internal helper function which sends a response */
+static void jingle_send_response(struct ast_xmpp_client *connection, ikspak *pak)
+{
+ iks *response;
+
+ if (!(response = iks_new("iq"))) {
+ ast_log(LOG_ERROR, "Unable to allocate an IKS response stanza\n");
+ return;
+ }
+
+ iks_insert_attrib(response, "type", "result");
+ iks_insert_attrib(response, "from", connection->jid->full);
+ iks_insert_attrib(response, "to", iks_find_attrib(pak->x, "from"));
+ iks_insert_attrib(response, "id", iks_find_attrib(pak->x, "id"));
+
+ ast_xmpp_client_send(connection, response);
+
+ iks_delete(response);
+}
+
+/*! \brief Internal helper function which sends an error response */
+static void jingle_send_error_response(struct ast_xmpp_client *connection, ikspak *pak, const char *type, const char *reasonstr, const char *reasonstr2)
+{
+ iks *response, *error = NULL, *reason = NULL, *reason2 = NULL;
+
+ if (!(response = iks_new("iq")) ||
+ !(error = iks_new("error")) ||
+ !(reason = iks_new(reasonstr))) {
+ ast_log(LOG_ERROR, "Unable to allocate IKS error response stanzas\n");
+ goto end;
+ }
+
+ iks_insert_attrib(response, "type", "error");
+ iks_insert_attrib(response, "from", connection->jid->full);
+ iks_insert_attrib(response, "to", iks_find_attrib(pak->x, "from"));
+ iks_insert_attrib(response, "id", iks_find_attrib(pak->x, "id"));
+
+ iks_insert_attrib(error, "type", type);
+ iks_insert_node(error, reason);
+
+ if (!ast_strlen_zero(reasonstr2) && (reason2 = iks_new(reasonstr2))) {
+ iks_insert_node(error, reason2);
+ }
+
+ iks_insert_node(response, error);
+
+ ast_xmpp_client_send(connection, response);
+end:
+ iks_delete(reason2);
+ iks_delete(reason);
+ iks_delete(error);
+ iks_delete(response);
+}
+
+/*! \brief Internal helper function which adds ICE-UDP candidates to a transport node */
+static int jingle_add_ice_udp_candidates_to_transport(struct ast_rtp_instance *rtp, iks *transport, iks **candidates, unsigned int maximum)
+{
+ struct ast_rtp_engine_ice *ice;
+ struct ao2_container *local_candidates;
+ struct ao2_iterator it;
+ struct ast_rtp_engine_ice_candidate *candidate;
+ int i = 0, res = 0;
+
+ if (!(ice = ast_rtp_instance_get_ice(rtp)) || !(local_candidates = ice->get_local_candidates(rtp))) {
+ ast_log(LOG_ERROR, "Unable to add ICE-UDP candidates as ICE support not available or no candidates available\n");
+ return -1;
+ }
+
+ iks_insert_attrib(transport, "xmlns", JINGLE_ICE_UDP_NS);
+ iks_insert_attrib(transport, "pwd", ice->get_password(rtp));
+ iks_insert_attrib(transport, "ufrag", ice->get_ufrag(rtp));
+
+ it = ao2_iterator_init(local_candidates, 0);
+
+ while ((candidate = ao2_iterator_next(&it)) && (i < maximum)) {
+ iks *local_candidate;
+ char tmp[30];
+
+ if (!(local_candidate = iks_new("candidate"))) {
+ res = -1;
+ ast_log(LOG_ERROR, "Unable to allocate IKS candidate stanza for ICE-UDP transport\n");
+ break;
+ }
+
+ snprintf(tmp, sizeof(tmp), "%d", candidate->id);
+ iks_insert_attrib(local_candidate, "component", tmp);
+ snprintf(tmp, sizeof(tmp), "%d", ast_str_hash(candidate->foundation));
+ iks_insert_attrib(local_candidate, "foundation", tmp);
+ iks_insert_attrib(local_candidate, "generation", "0");
+ snprintf(tmp, sizeof(tmp), "%04lx", ast_random() & 0xffff);
+ iks_insert_attrib(local_candidate, "id", tmp);
+ iks_insert_attrib(local_candidate, "ip", ast_sockaddr_stringify_host(&candidate->address));
+ iks_insert_attrib(local_candidate, "port", ast_sockaddr_stringify_port(&candidate->address));
+ snprintf(tmp, sizeof(tmp), "%d", candidate->priority);
+ iks_insert_attrib(local_candidate, "priority", tmp);
+ iks_insert_attrib(local_candidate, "protocol", "udp");
+
+ if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
+ iks_insert_attrib(local_candidate, "type", "host");
+ } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
+ iks_insert_attrib(local_candidate, "type", "srflx");
+ } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_RELAYED) {
+ iks_insert_attrib(local_candidate, "type", "relay");
+ }
+
+ iks_insert_node(transport, local_candidate);
+ candidates[i++] = local_candidate;
+ }
+
+ ao2_iterator_destroy(&it);
+ ao2_ref(local_candidates, -1);
+
+ return res;
+}
+
+/*! \brief Internal helper function which adds Google candidates to a transport node */
+static int jingle_add_google_candidates_to_transport(struct ast_rtp_instance *rtp, iks *transport, iks **candidates, unsigned int video, enum jingle_transport transport_type, unsigned int maximum)
+{
+ struct ast_rtp_engine_ice *ice;
+ struct ao2_container *local_candidates;
+ struct ao2_iterator it;
+ struct ast_rtp_engine_ice_candidate *candidate;
+ int i = 0, res = 0;
+
+ if (!(ice = ast_rtp_instance_get_ice(rtp)) || !(local_candidates = ice->get_local_candidates(rtp))) {
+ ast_log(LOG_ERROR, "Unable to add Google ICE candidates as ICE support not available or no candidates available\n");
+ return -1;
+ }
+
+ if (transport_type != JINGLE_TRANSPORT_GOOGLE_V1) {
+ iks_insert_attrib(transport, "xmlns", GOOGLE_TRANSPORT_NS);
+ }
+
+ it = ao2_iterator_init(local_candidates, 0);
+
+ while ((candidate = ao2_iterator_next(&it)) && (i < maximum)) {
+ iks *local_candidate;
+ /* In Google land a username is 16 bytes, explicitly */
+ char ufrag[17] = "";
+
+ if (!(local_candidate = iks_new("candidate"))) {
+ res = -1;
+ ast_log(LOG_ERROR, "Unable to allocate IKS candidate stanza for Google ICE transport\n");
+ break;
+ }
+
+ /* We only support RTP candidates */
+ if (candidate->id != 1) {
+ continue;
+ }
+
+ iks_insert_attrib(local_candidate, "name", !video ? "rtp" : "video_rtp");
+ iks_insert_attrib(local_candidate, "address", ast_sockaddr_stringify_host(&candidate->address));
+ iks_insert_attrib(local_candidate, "port", ast_sockaddr_stringify_port(&candidate->address));
+
+ if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
+ iks_insert_attrib(local_candidate, "preference", "0.95");
+ iks_insert_attrib(local_candidate, "type", "local");
+ } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
+ iks_insert_attrib(local_candidate, "preference", "0.9");
+ iks_insert_attrib(local_candidate, "type", "stun");
+ }
+
+ iks_insert_attrib(local_candidate, "protocol", "udp");
+ iks_insert_attrib(local_candidate, "network", "0");
+ snprintf(ufrag, sizeof(ufrag), "%s", ice->get_ufrag(rtp));
+ iks_insert_attrib(local_candidate, "username", ufrag);
+ iks_insert_attrib(local_candidate, "generation", "0");
+
+ if (transport_type == JINGLE_TRANSPORT_GOOGLE_V1) {
+ iks_insert_attrib(local_candidate, "password", "");
+ iks_insert_attrib(local_candidate, "foundation", "0");
+ iks_insert_attrib(local_candidate, "component", "1");
+ } else {
+ iks_insert_attrib(local_candidate, "password", ice->get_password(rtp));
+ }
+
+ /* You may notice a lack of relay support up above - this is because we don't support it for use with
+ * the Google talk transport due to their arcane support. */
+
+ iks_insert_node(transport, local_candidate);
+ candidates[i++] = local_candidate;
+ }
+
+ ao2_iterator_destroy(&it);
+ ao2_ref(local_candidates, -1);
+
+ return res;
+}
+
+/*! \brief Internal function which sends a session-terminate message */
+static void jingle_send_session_terminate(struct jingle_session *session, const char *reasontext)
+{
+ iks *iq = NULL, *jingle = NULL, *reason = NULL, *text = NULL;
+
+ if (!(iq = iks_new("iq")) || !(jingle = iks_new(session->transport == JINGLE_TRANSPORT_GOOGLE_V1 ? "session" : "jingle")) ||
+ !(reason = iks_new("reason")) || !(text = iks_new(reasontext))) {
+ ast_log(LOG_ERROR, "Failed to allocate stanzas for session-terminate message on session '%s'\n", session->sid);
+ goto end;
+ }
+
+ iks_insert_attrib(iq, "to", session->remote);
+ iks_insert_attrib(iq, "type", "set");
+ iks_insert_attrib(iq, "id", session->connection->mid);
+ ast_xmpp_increment_mid(session->connection->mid);
+
+ if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
+ iks_insert_attrib(jingle, "type", "terminate");
+ iks_insert_attrib(jingle, "id", session->sid);
+ iks_insert_attrib(jingle, "xmlns", GOOGLE_SESSION_NS);
+ iks_insert_attrib(jingle, "initiator", session->outgoing ? session->connection->jid->full : session->remote);
+ } else {
+ iks_insert_attrib(jingle, "action", "session-terminate");
+ iks_insert_attrib(jingle, "sid", session->sid);
+ iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
+ }
+
+ iks_insert_node(iq, jingle);
+ iks_insert_node(jingle, reason);
+ iks_insert_node(reason, text);
+
+ ast_xmpp_client_send(session->connection, iq);
+
+end:
+ iks_delete(text);
+ iks_delete(reason);
+ iks_delete(jingle);
+ iks_delete(iq);
+}
+
+/*! \brief Internal function which sends a session-info message */
+static void jingle_send_session_info(struct jingle_session *session, const char *info)
+{
+ iks *iq = NULL, *jingle = NULL, *text = NULL;
+
+ /* Google-V1 has no way to send informational messages so don't even bother trying */
+ if (session->transport == JINGLE_TRANSPORT_GOOGLE_V1) {
+ return;
+ }
+
+ if (!(iq = iks_new("iq")) || !(jingle = iks_new("jingle")) || !(text = iks_new(info))) {
+ ast_log(LOG_ERROR, "Failed to allocate stanzas for session-info message on session '%s'\n", session->sid);
+ goto end;
+ }
+
+ iks_insert_attrib(iq, "to", session->remote);
+ iks_insert_attrib(iq, "type", "set");
+ iks_insert_attrib(iq, "id", session->connection->mid);
+ ast_xmpp_increment_mid(session->connection->mid);
+
+ iks_insert_attrib(jingle, "action", "session-info");
+ iks_insert_attrib(jingle, "sid", session->sid);
+ iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
+ iks_insert_node(iq, jingle);
+ iks_insert_node(jingle, text);
+
+ ast_xmpp_client_send(session->connection, iq);
+
+end:
+ iks_delete(text);
+ iks_delete(jingle);
+ iks_delete(iq);
+}
+
+/*! \internal
+ *
+ * \brief Locks both pvt and pvt owner if owner is present.
+ *
+ * \note This function gives a ref to pvt->owner if it is present and locked.
+ * This reference must be decremented after pvt->owner is unlocked.
+ *
+ * \note This function will never give you up,
+ * \note This function will never let you down.
+ * \note This function will run around and desert you.
+ *
+ * \pre pvt is not locked
+ * \post pvt is locked
+ * \post pvt->owner is locked and its reference count is increased (if pvt->owner is not NULL)
+ *
+ * \returns a pointer to the locked and reffed pvt->owner channel if it exists.
+ */
+static struct ast_channel *jingle_session_lock_full(struct jingle_session *pvt)
+{
+ struct ast_channel *chan;
+
+ /* Locking is simple when it is done right. If you see a deadlock resulting
+ * in this function, it is not this function's fault, Your problem exists elsewhere.
+ * This function is perfect... seriously. */
+ for (;;) {
+ /* First, get the channel and grab a reference to it */
+ ao2_lock(pvt);
+ chan = pvt->owner;
+ if (chan) {
+ /* The channel can not go away while we hold the pvt lock.
+ * Give the channel a ref so it will not go away after we let
+ * the pvt lock go. */
+ ast_channel_ref(chan);
+ } else {
+ /* no channel, return pvt locked */
+ return NULL;
+ }
+
+ /* We had to hold the pvt lock while getting a ref to the owner channel
+ * but now we have to let this lock go in order to preserve proper
+ * locking order when grabbing the channel lock */
+ ao2_unlock(pvt);
+
+ /* Look, no deadlock avoidance, hooray! */
+ ast_channel_lock(chan);
+ ao2_lock(pvt);
+ if (pvt->owner == chan) {
+ /* done */
+ break;
+ }
+
+ /* If the owner changed while everything was unlocked, no problem,
+ * just start over and everthing will work. This is rare, do not be
+ * confused by this loop and think this it is an expensive operation.
+ * The majority of the calls to this function will never involve multiple
+ * executions of this loop. */
[... 1724 lines stripped ...]
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