[asterisk-commits] mjordan: branch 12 r403618 - in /branches/12: channels/ channels/pjsip/ chann...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Dec 11 07:05:20 CST 2013
Author: mjordan
Date: Wed Dec 11 07:05:12 2013
New Revision: 403618
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=403618
Log:
func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
Added:
branches/12/channels/pjsip/
branches/12/channels/pjsip/dialplan_functions.c (with props)
branches/12/channels/pjsip/include/
branches/12/channels/pjsip/include/chan_pjsip.h (with props)
branches/12/channels/pjsip/include/dialplan_functions.h (with props)
Modified:
branches/12/channels/Makefile
branches/12/channels/chan_pjsip.c
branches/12/funcs/func_channel.c
branches/12/include/asterisk/res_pjsip_session.h
branches/12/main/xmldoc.c
branches/12/res/res_pjsip_t38.c
Modified: branches/12/channels/Makefile
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/Makefile?view=diff&rev=403618&r1=403617&r2=403618
==============================================================================
--- branches/12/channels/Makefile (original)
+++ branches/12/channels/Makefile Wed Dec 11 07:05:12 2013
@@ -77,6 +77,9 @@
$(if $(filter chan_sip,$(EMBEDDED_MODS)),modules.link,chan_sip.so): $(subst .c,.o,$(wildcard sip/*.c))
$(subst .c,.o,$(wildcard sip/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_sip)
+$(if $(filter chan_pjsip,$(EMBEDDED_MODS)),modules.link,chan_pjsip.so): $(subst .c,.o,$(wildcard pjsip/*.c))
+$(subst .c,.o,$(wildcard pjsip/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_pjsip)
+
# Additional objects to combine with chan_dahdi.so
CHAN_DAHDI_OBJS= \
$(subst .c,.o,$(wildcard dahdi/*.c)) \
Modified: branches/12/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_pjsip.c?view=diff&rev=403618&r1=403617&r2=403618
==============================================================================
--- branches/12/channels/chan_pjsip.c (original)
+++ branches/12/channels/chan_pjsip.c Wed Dec 11 07:05:12 2013
@@ -61,61 +61,13 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
-/*** DOCUMENTATION
- <function name="PJSIP_DIAL_CONTACTS" language="en_US">
- <synopsis>
- Return a dial string for dialing all contacts on an AOR.
- </synopsis>
- <syntax>
- <parameter name="endpoint" required="true">
- <para>Name of the endpoint</para>
- </parameter>
- <parameter name="aor" required="false">
- <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
- </parameter>
- <parameter name="request_user" required="false">
- <para>Optional request user to use in the request URI</para>
- </parameter>
- </syntax>
- <description>
- <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
- </description>
- </function>
- <function name="PJSIP_MEDIA_OFFER" language="en_US">
- <synopsis>
- Media and codec offerings to be set on an outbound SIP channel prior to dialing.
- </synopsis>
- <syntax>
- <parameter name="media" required="true">
- <para>types of media offered</para>
- </parameter>
- </syntax>
- <description>
- <para>Returns the codecs offered based upon the media choice</para>
- </description>
- </function>
- ***/
+#include "pjsip/include/chan_pjsip.h"
+#include "pjsip/include/dialplan_functions.h"
static const char desc[] = "PJSIP Channel";
static const char channel_type[] = "PJSIP";
static unsigned int chan_idx;
-
-/*!
- * \brief Positions of various media
- */
-enum sip_session_media_position {
- /*! \brief First is audio */
- SIP_MEDIA_AUDIO = 0,
- /*! \brief Second is video */
- SIP_MEDIA_VIDEO,
- /*! \brief Last is the size for media details */
- SIP_MEDIA_SIZE,
-};
-
-struct chan_pjsip_pvt {
- struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
-};
static void chan_pjsip_pvt_dtor(void *obj)
{
@@ -145,7 +97,7 @@
static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
/*! \brief PBX interface structure for channel registration */
-static struct ast_channel_tech chan_pjsip_tech = {
+struct ast_channel_tech chan_pjsip_tech = {
.type = channel_type,
.description = "PJSIP Channel Driver",
.requester = chan_pjsip_request,
@@ -164,6 +116,7 @@
.fixup = chan_pjsip_fixup,
.devicestate = chan_pjsip_devicestate,
.queryoption = chan_pjsip_queryoption,
+ .func_channel_read = pjsip_acf_channel_read,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
@@ -191,184 +144,6 @@
.incoming_request = chan_pjsip_incoming_ack,
};
-/*! \brief Dialplan function for constructing a dial string for calling all contacts */
-static int chan_pjsip_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
-{
- RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
- RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
- const char *aor_name;
- char *rest;
-
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(endpoint_name);
- AST_APP_ARG(aor_name);
- AST_APP_ARG(request_user);
- );
-
- AST_STANDARD_APP_ARGS(args, data);
-
- if (ast_strlen_zero(args.endpoint_name)) {
- ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
- return -1;
- } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
- ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
- return -1;
- }
-
- aor_name = S_OR(args.aor_name, endpoint->aors);
-
- if (ast_strlen_zero(aor_name)) {
- ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
- return -1;
- } else if (!(dial = ast_str_create(len))) {
- ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
- return -1;
- } else if (!(rest = ast_strdupa(aor_name))) {
- ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
- return -1;
- }
-
- while ((aor_name = strsep(&rest, ","))) {
- RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
- RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
- struct ao2_iterator it_contacts;
- struct ast_sip_contact *contact;
-
- if (!aor) {
- /* If the AOR provided is not found skip it, there may be more */
- continue;
- } else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
- /* No contacts are available, skip it as well */
- continue;
- } else if (!ao2_container_count(contacts)) {
- /* We were given a container but no contacts are in it... */
- continue;
- }
-
- it_contacts = ao2_iterator_init(contacts, 0);
- for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
- ast_str_append(&dial, -1, "PJSIP/");
-
- if (!ast_strlen_zero(args.request_user)) {
- ast_str_append(&dial, -1, "%s@", args.request_user);
- }
- ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
- }
- ao2_iterator_destroy(&it_contacts);
- }
-
- /* Trim the '&' at the end off */
- ast_str_truncate(dial, ast_str_strlen(dial) - 1);
-
- ast_copy_string(buf, ast_str_buffer(dial), len);
-
- return 0;
-}
-
-static struct ast_custom_function chan_pjsip_dial_contacts_function = {
- .name = "PJSIP_DIAL_CONTACTS",
- .read = chan_pjsip_dial_contacts,
-};
-
-static int media_offer_read_av(struct ast_sip_session *session, char *buf,
- size_t len, enum ast_format_type media_type)
-{
- int i, size = 0;
- struct ast_format fmt;
- const char *name;
-
- for (i = 0; ast_codec_pref_index(&session->override_prefs, i, &fmt); ++i) {
- if (AST_FORMAT_GET_TYPE(fmt.id) != media_type) {
- continue;
- }
-
- name = ast_getformatname(&fmt);
-
- if (ast_strlen_zero(name)) {
- ast_log(LOG_WARNING, "PJSIP_MEDIA_OFFER unrecognized format %s\n", name);
- continue;
- }
-
- /* add one since we'll include a comma */
- size = strlen(name) + 1;
- len -= size;
- if ((len) < 0) {
- break;
- }
-
- /* no reason to use strncat here since we have already ensured buf has
- enough space, so strcat can be safely used */
- strcat(buf, name);
- strcat(buf, ",");
- }
-
- if (size) {
- /* remove the extra comma */
- buf[strlen(buf) - 1] = '\0';
- }
- return 0;
-}
-
-struct media_offer_data {
- struct ast_sip_session *session;
- enum ast_format_type media_type;
- const char *value;
-};
-
-static int media_offer_write_av(void *obj)
-{
- struct media_offer_data *data = obj;
- int i;
- struct ast_format fmt;
- /* remove all of the given media type first */
- for (i = 0; ast_codec_pref_index(&data->session->override_prefs, i, &fmt); ++i) {
- if (AST_FORMAT_GET_TYPE(fmt.id) == data->media_type) {
- ast_codec_pref_remove(&data->session->override_prefs, &fmt);
- }
- }
- ast_format_cap_remove_bytype(data->session->req_caps, data->media_type);
- ast_parse_allow_disallow(&data->session->override_prefs, data->session->req_caps, data->value, 1);
-
- return 0;
-}
-
-static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
-{
- struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
-
- if (!strcmp(data, "audio")) {
- return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
- } else if (!strcmp(data, "video")) {
- return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
- }
-
- return 0;
-}
-
-static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
-{
- struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
-
- struct media_offer_data mdata = {
- .session = channel->session,
- .value = value
- };
-
- if (!strcmp(data, "audio")) {
- mdata.media_type = AST_FORMAT_TYPE_AUDIO;
- } else if (!strcmp(data, "video")) {
- mdata.media_type = AST_FORMAT_TYPE_VIDEO;
- }
-
- return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
-}
-
-static struct ast_custom_function media_offer_function = {
- .name = "PJSIP_MEDIA_OFFER",
- .read = media_offer_read,
- .write = media_offer_write
-};
-
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
@@ -436,6 +211,20 @@
return ast_sip_session_refresh(session, NULL, NULL, NULL,
session->endpoint->media.direct_media.method, 1);
}
+
+/*! \brief Destructor function for \ref transport_info_data */
+static void transport_info_destroy(void *obj)
+{
+ struct transport_info_data *data = obj;
+ ast_free(data);
+}
+
+/*! \brief Datastore used to store local/remote addresses for the
+ * INVITE request that created the PJSIP channel */
+static struct ast_datastore_info transport_info = {
+ .type = "chan_pjsip_transport_info",
+ .destroy = transport_info_destroy,
+};
static struct ast_datastore_info direct_media_mitigation_info = { };
@@ -1989,11 +1778,27 @@
/*! \brief Function called when a request is received on the session */
static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
+ RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
+ struct transport_info_data *transport_data;
pjsip_tx_data *packet = NULL;
if (session->channel) {
return 0;
}
+
+ datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
+ if (!datastore) {
+ return -1;
+ }
+
+ transport_data = ast_calloc(1, sizeof(*transport_data));
+ if (!transport_data) {
+ return -1;
+ }
+ pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
+ pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
+ datastore->data = transport_data;
+ ast_sip_session_add_datastore(session, datastore);
if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
@@ -2077,6 +1882,17 @@
}
return 0;
}
+
+static struct ast_custom_function chan_pjsip_dial_contacts_function = {
+ .name = "PJSIP_DIAL_CONTACTS",
+ .read = pjsip_acf_dial_contacts_read,
+};
+
+static struct ast_custom_function media_offer_function = {
+ .name = "PJSIP_MEDIA_OFFER",
+ .read = pjsip_acf_media_offer_read,
+ .write = pjsip_acf_media_offer_write
+};
/*!
* \brief Load the module
@@ -2110,6 +1926,7 @@
if (ast_custom_function_register(&media_offer_function)) {
ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
+ goto end;
}
if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
@@ -2150,13 +1967,13 @@
/*! \brief Unload the PJSIP channel from Asterisk */
static int unload_module(void)
{
- ast_custom_function_unregister(&media_offer_function);
-
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
+ ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
+
ast_channel_unregister(&chan_pjsip_tech);
ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
Added: branches/12/channels/pjsip/dialplan_functions.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/pjsip/dialplan_functions.c?view=auto&rev=403618
==============================================================================
--- branches/12/channels/pjsip/dialplan_functions.c (added)
+++ branches/12/channels/pjsip/dialplan_functions.c Wed Dec 11 07:05:12 2013
@@ -1,0 +1,893 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * 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 \verbatim Joshua Colp <jcolp at digium.com> \endverbatim
+ * \author \verbatim Matt Jordan <mjordan at digium.com> \endverbatim
+ *
+ * \ingroup functions
+ *
+ * \brief PJSIP channel dialplan functions
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+/*** DOCUMENTATION
+<function name="PJSIP_DIAL_CONTACTS" language="en_US">
+ <synopsis>
+ Return a dial string for dialing all contacts on an AOR.
+ </synopsis>
+ <syntax>
+ <parameter name="endpoint" required="true">
+ <para>Name of the endpoint</para>
+ </parameter>
+ <parameter name="aor" required="false">
+ <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
+ </parameter>
+ <parameter name="request_user" required="false">
+ <para>Optional request user to use in the request URI</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
+ </description>
+</function>
+<function name="PJSIP_MEDIA_OFFER" language="en_US">
+ <synopsis>
+ Media and codec offerings to be set on an outbound SIP channel prior to dialing.
+ </synopsis>
+ <syntax>
+ <parameter name="media" required="true">
+ <para>types of media offered</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Returns the codecs offered based upon the media choice</para>
+ </description>
+</function>
+<info name="PJSIPCHANNEL" language="en_US" tech="PJSIP">
+ <enumlist>
+ <enum name="rtp">
+ <para>R/O Retrieve media related information.</para>
+ <parameter name="type" required="true">
+ <para>When <replaceable>rtp</replaceable> is specified, the
+ <literal>type</literal> parameter must be provided. It specifies
+ which RTP parameter to read.</para>
+ <enumlist>
+ <enum name="src">
+ <para>Retrieve the local address for RTP.</para>
+ </enum>
+ <enum name="dest">
+ <para>Retrieve the remote address for RTP.</para>
+ </enum>
+ <enum name="direct">
+ <para>If direct media is enabled, this address is the remote address
+ used for RTP.</para>
+ </enum>
+ <enum name="secure">
+ <para>Whether or not the media stream is encrypted.</para>
+ <enumlist>
+ <enum name="0">
+ <para>The media stream is not encrypted.</para>
+ </enum>
+ <enum name="1">
+ <para>The media stream is encrypted.</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="hold">
+ <para>Whether or not the media stream is currently restricted
+ due to a call hold.</para>
+ <enumlist>
+ <enum name="0">
+ <para>The media stream is not held.</para>
+ </enum>
+ <enum name="1">
+ <para>The media stream is held.</para>
+ </enum>
+ </enumlist>
+ </enum>
+ </enumlist>
+ </parameter>
+ <parameter name="media_type" required="false">
+ <para>When <replaceable>rtp</replaceable> is specified, the
+ <literal>media_type</literal> parameter may be provided. It specifies
+ which media stream the chosen RTP parameter should be retrieved
+ from.</para>
+ <enumlist>
+ <enum name="audio">
+ <para>Retrieve information from the audio media stream.</para>
+ <note><para>If not specified, <literal>audio</literal> is used
+ by default.</para></note>
+ </enum>
+ <enum name="video">
+ <para>Retrieve information from the video media stream.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </enum>
+ <enum name="rtcp">
+ <para>R/O Retrieve RTCP statistics.</para>
+ <parameter name="statistic" required="true">
+ <para>When <replaceable>rtcp</replaceable> is specified, the
+ <literal>statistic</literal> parameter must be provided. It specifies
+ which RTCP statistic parameter to read.</para>
+ <enumlist>
+ <enum name="all">
+ <para>Retrieve a summary of all RTCP statistics.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="ssrc">
+ <para>Our Synchronization Source identifier</para>
+ </enum>
+ <enum name="themssrc">
+ <para>Their Synchronization Source identifier</para>
+ </enum>
+ <enum name="lp">
+ <para>Our lost packet count</para>
+ </enum>
+ <enum name="rxjitter">
+ <para>Received packet jitter</para>
+ </enum>
+ <enum name="rxcount">
+ <para>Received packet count</para>
+ </enum>
+ <enum name="txjitter">
+ <para>Transmitted packet jitter</para>
+ </enum>
+ <enum name="txcount">
+ <para>Transmitted packet count</para>
+ </enum>
+ <enum name="rlp">
+ <para>Remote lost packet count</para>
+ </enum>
+ <enum name="rtt">
+ <para>Round trip time</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_jitter">
+ <para>Retrieve a summary of all RTCP Jitter statistics.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minrxjitter">
+ <para>Our minimum jitter</para>
+ </enum>
+ <enum name="maxrxjitter">
+ <para>Our max jitter</para>
+ </enum>
+ <enum name="avgrxjitter">
+ <para>Our average jitter</para>
+ </enum>
+ <enum name="stdevrxjitter">
+ <para>Our jitter standard deviation</para>
+ </enum>
+ <enum name="reported_minjitter">
+ <para>Their minimum jitter</para>
+ </enum>
+ <enum name="reported_maxjitter">
+ <para>Their max jitter</para>
+ </enum>
+ <enum name="reported_avgjitter">
+ <para>Their average jitter</para>
+ </enum>
+ <enum name="reported_stdevjitter">
+ <para>Their jitter standard deviation</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_loss">
+ <para>Retrieve a summary of all RTCP packet loss statistics.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minrxlost">
+ <para>Our minimum lost packets</para>
+ </enum>
+ <enum name="maxrxlost">
+ <para>Our max lost packets</para>
+ </enum>
+ <enum name="avgrxlost">
+ <para>Our average lost packets</para>
+ </enum>
+ <enum name="stdevrxlost">
+ <para>Our lost packets standard deviation</para>
+ </enum>
+ <enum name="reported_minlost">
+ <para>Their minimum lost packets</para>
+ </enum>
+ <enum name="reported_maxlost">
+ <para>Their max lost packets</para>
+ </enum>
+ <enum name="reported_avglost">
+ <para>Their average lost packets</para>
+ </enum>
+ <enum name="reported_stdevlost">
+ <para>Their lost packets standard deviation</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_rtt">
+ <para>Retrieve a summary of all RTCP round trip time information.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minrtt">
+ <para>Minimum round trip time</para>
+ </enum>
+ <enum name="maxrtt">
+ <para>Maximum round trip time</para>
+ </enum>
+ <enum name="avgrtt">
+ <para>Average round trip time</para>
+ </enum>
+ <enum name="stdevrtt">
+ <para>Standard deviation round trip time</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="txcount"><para>Transmitted packet count</para></enum>
+ <enum name="rxcount"><para>Received packet count</para></enum>
+ <enum name="txjitter"><para>Transmitted packet jitter</para></enum>
+ <enum name="rxjitter"><para>Received packet jitter</para></enum>
+ <enum name="remote_maxjitter"><para>Their max jitter</para></enum>
+ <enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
+ <enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
+ <enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
+ <enum name="local_maxjitter"><para>Our max jitter</para></enum>
+ <enum name="local_minjitter"><para>Our minimum jitter</para></enum>
+ <enum name="local_normdevjitter"><para>Our average jitter</para></enum>
+ <enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
+ <enum name="txploss"><para>Transmitted packet loss</para></enum>
+ <enum name="rxploss"><para>Received packet loss</para></enum>
+ <enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
+ <enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
+ <enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
+ <enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
+ <enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
+ <enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
+ <enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
+ <enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
+ <enum name="rtt"><para>Round trip time</para></enum>
+ <enum name="maxrtt"><para>Maximum round trip time</para></enum>
+ <enum name="minrtt"><para>Minimum round trip time</para></enum>
+ <enum name="normdevrtt"><para>Average round trip time</para></enum>
+ <enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
+ <enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
+ <enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
+ </enumlist>
+ </parameter>
+ <parameter name="media_type" required="false">
+ <para>When <replaceable>rtcp</replaceable> is specified, the
+ <literal>media_type</literal> parameter may be provided. It specifies
+ which media stream the chosen RTCP parameter should be retrieved
+ from.</para>
+ <enumlist>
+ <enum name="audio">
+ <para>Retrieve information from the audio media stream.</para>
+ <note><para>If not specified, <literal>audio</literal> is used
+ by default.</para></note>
+ </enum>
+ <enum name="video">
+ <para>Retrieve information from the video media stream.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </enum>
+ <enum name="endpoint">
+ <para>R/O The name of the endpoint associated with this channel.
+ Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
+ further endpoint related information.</para>
+ </enum>
+ <enum name="pjsip">
+ <para>R/O Obtain information about the current PJSIP channel and its
+ session.</para>
+ <parameter name="type" required="true">
+ <para>When <replaceable>pjsip</replaceable> is specified, the
+ <literal>type</literal> parameter must be provided. It specifies
+ which signalling parameter to read.</para>
+ <enumlist>
+ <enum name="secure">
+ <para>Whether or not the signalling uses a secure transport.</para>
+ <enumlist>
+ <enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
+ <enum name="1"><para>The signalling uses a secure transport.</para></enum>
+ </enumlist>
+ </enum>
+ <enum name="target_uri">
+ <para>The request URI of the <literal>INVITE</literal> request associated with the creation of this channel.</para>
+ </enum>
+ <enum name="local_uri">
+ <para>The local URI.</para>
+ </enum>
+ <enum name="remote_uri">
+ <para>The remote URI.</para>
+ </enum>
+ <enum name="t38state">
+ <para>The current state of any T.38 fax on this channel.</para>
+ <enumlist>
+ <enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
+ <enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
+ <enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
+ <enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
+ <enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
+ </enumlist>
+ </enum>
+ <enum name="local_addr">
+ <para>On inbound calls, the full IP address and port number that
+ the <literal>INVITE</literal> request was received on. On outbound
+ calls, the full IP address and port number that the <literal>INVITE</literal>
+ request was transmitted from.</para>
+ </enum>
+ <enum name="remote_addr">
+ <para>On inbound calls, the full IP address and port number that
+ the <literal>INVITE</literal> request was received from. On outbound
+ calls, the full IP address and port number that the <literal>INVITE</literal>
+ request was transmitted to.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </enum>
+ </enumlist>
+</info>
+***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjlib.h>
+#include <pjsip_ua.h>
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/astobj2.h"
+#include "asterisk/module.h"
+#include "asterisk/acl.h"
+#include "asterisk/app.h"
+#include "asterisk/channel.h"
+#include "asterisk/format.h"
+#include "asterisk/pbx.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+#include "include/chan_pjsip.h"
+#include "include/dialplan_functions.h"
+
+/*!
+ * \brief String representations of the T.38 state enum
+ */
+static const char *t38state_to_string[T38_MAX_ENUM] = {
+ [T38_DISABLED] = "DISABLED",
+ [T38_LOCAL_REINVITE] = "LOCAL_REINVITE",
+ [T38_PEER_REINVITE] = "REMOTE_REINVITE",
+ [T38_ENABLED] = "ENABLED",
+ [T38_REJECTED] = "REJECTED",
+};
+
+/*!
+ * \internal \brief Handle reading RTP information
+ */
+static int channel_read_rtp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
+{
+ struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+ struct chan_pjsip_pvt *pvt;
+ struct ast_sip_session_media *media = NULL;
+ struct ast_sockaddr addr;
+
+ if (!channel) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+ return -1;
+ }
+
+ pvt = channel->pvt;
+ if (!pvt) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no chan_pjsip pvt!\n", ast_channel_name(chan));
+ return -1;
+ }
+
+ if (ast_strlen_zero(type)) {
+ ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtp' information\n");
+ return -1;
+ }
+
+ if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
+ media = pvt->media[SIP_MEDIA_AUDIO];
+ } else if (!strcmp(field, "video")) {
+ media = pvt->media[SIP_MEDIA_VIDEO];
+ } else {
+ ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtp' information\n", field);
+ return -1;
+ }
+
+ if (!media || !media->rtp) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
+ ast_channel_name(chan), S_OR(field, "audio"));
+ return -1;
+ }
+
+ if (!strcmp(type, "src")) {
+ ast_rtp_instance_get_local_address(media->rtp, &addr);
+ ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
+ } else if (!strcmp(type, "dest")) {
+ ast_rtp_instance_get_remote_address(media->rtp, &addr);
+ ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
+ } else if (!strcmp(type, "direct")) {
+ ast_copy_string(buf, ast_sockaddr_stringify(&media->direct_media_addr), buflen);
+ } else if (!strcmp(type, "secure")) {
+ snprintf(buf, buflen, "%u", media->srtp ? 1 : 0);
+ } else if (!strcmp(type, "hold")) {
+ snprintf(buf, buflen, "%u", media->held ? 1 : 0);
+ } else {
+ ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type);
+ return -1;
+ }
+
+ return 0;
+}
+
+/*!
+ * \internal \brief Handle reading RTCP information
+ */
+static int channel_read_rtcp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
+{
+ struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+ struct chan_pjsip_pvt *pvt;
+ struct ast_sip_session_media *media = NULL;
+
+ if (!channel) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+ return -1;
+ }
+
+ pvt = channel->pvt;
+ if (!pvt) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no chan_pjsip pvt!\n", ast_channel_name(chan));
+ return -1;
+ }
+
+ if (ast_strlen_zero(type)) {
+ ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtcp' information\n");
+ return -1;
+ }
+
+ if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
+ media = pvt->media[SIP_MEDIA_AUDIO];
+ } else if (!strcmp(field, "video")) {
+ media = pvt->media[SIP_MEDIA_VIDEO];
+ } else {
+ ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtcp' information\n", field);
+ return -1;
+ }
+
+ if (!media || !media->rtp) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
+ ast_channel_name(chan), S_OR(field, "audio"));
+ return -1;
+ }
+
+ if (!strncasecmp(type, "all", 3)) {
+ enum ast_rtp_instance_stat_field stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY;
+
+ if (!strcasecmp(type, "all_jitter")) {
+ stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER;
+ } else if (!strcasecmp(type, "all_rtt")) {
+ stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT;
+ } else if (!strcasecmp(type, "all_loss")) {
+ stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS;
+ }
+
+ if (!ast_rtp_instance_get_quality(media->rtp, stat_field, buf, buflen)) {
+ ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
+ return -1;
+ }
+ } else {
+ struct ast_rtp_instance_stats stats;
+ int i;
+ struct {
+ const char *name;
+ enum { INT, DBL } type;
+ union {
+ unsigned int *i4;
+ double *d8;
+ };
+ } lookup[] = {
+ { "txcount", INT, { .i4 = &stats.txcount, }, },
+ { "rxcount", INT, { .i4 = &stats.rxcount, }, },
+ { "txjitter", DBL, { .d8 = &stats.txjitter, }, },
+ { "rxjitter", DBL, { .d8 = &stats.rxjitter, }, },
+ { "remote_maxjitter", DBL, { .d8 = &stats.remote_maxjitter, }, },
+ { "remote_minjitter", DBL, { .d8 = &stats.remote_minjitter, }, },
+ { "remote_normdevjitter", DBL, { .d8 = &stats.remote_normdevjitter, }, },
+ { "remote_stdevjitter", DBL, { .d8 = &stats.remote_stdevjitter, }, },
+ { "local_maxjitter", DBL, { .d8 = &stats.local_maxjitter, }, },
+ { "local_minjitter", DBL, { .d8 = &stats.local_minjitter, }, },
+ { "local_normdevjitter", DBL, { .d8 = &stats.local_normdevjitter, }, },
+ { "local_stdevjitter", DBL, { .d8 = &stats.local_stdevjitter, }, },
+ { "txploss", INT, { .i4 = &stats.txploss, }, },
+ { "rxploss", INT, { .i4 = &stats.rxploss, }, },
+ { "remote_maxrxploss", DBL, { .d8 = &stats.remote_maxrxploss, }, },
+ { "remote_minrxploss", DBL, { .d8 = &stats.remote_minrxploss, }, },
+ { "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
+ { "remote_stdevrxploss", DBL, { .d8 = &stats.remote_stdevrxploss, }, },
+ { "local_maxrxploss", DBL, { .d8 = &stats.local_maxrxploss, }, },
+ { "local_minrxploss", DBL, { .d8 = &stats.local_minrxploss, }, },
+ { "local_normdevrxploss", DBL, { .d8 = &stats.local_normdevrxploss, }, },
+ { "local_stdevrxploss", DBL, { .d8 = &stats.local_stdevrxploss, }, },
+ { "rtt", DBL, { .d8 = &stats.rtt, }, },
+ { "maxrtt", DBL, { .d8 = &stats.maxrtt, }, },
+ { "minrtt", DBL, { .d8 = &stats.minrtt, }, },
+ { "normdevrtt", DBL, { .d8 = &stats.normdevrtt, }, },
+ { "stdevrtt", DBL, { .d8 = &stats.stdevrtt, }, },
+ { "local_ssrc", INT, { .i4 = &stats.local_ssrc, }, },
+ { "remote_ssrc", INT, { .i4 = &stats.remote_ssrc, }, },
+ { NULL, },
+ };
+
+ if (ast_rtp_instance_get_stats(media->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+ ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
+ return -1;
+ }
+
+ for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
+ if (!strcasecmp(type, lookup[i].name)) {
+ if (lookup[i].type == INT) {
+ snprintf(buf, buflen, "%u", *lookup[i].i4);
+ } else {
+ snprintf(buf, buflen, "%f", *lookup[i].d8);
+ }
+ return 0;
+ }
+ }
+ ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'rtcp' information\n", type);
+ return -1;
+ }
+
+ return 0;
+}
+
+/*!
+ * \internal \brief Handle reading signalling information
+ */
+static int channel_read_pjsip(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
+{
+ struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+ char *buf_copy;
+ pjsip_dialog *dlg;
+
+ if (!channel) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+ return -1;
+ }
+
+ dlg = channel->session->inv_session->dlg;
+
+ if (!strcmp(type, "secure")) {
+ snprintf(buf, buflen, "%u", dlg->secure ? 1 : 0);
+ } else if (!strcmp(type, "target_uri")) {
+ pjsip_uri_print(PJSIP_URI_IN_REQ_URI, dlg->target, buf, sizeof(buflen));
+ buf_copy = ast_strdupa(buf);
+ ast_escape_quoted(buf_copy, buf, buflen);
+ } else if (!strcmp(type, "local_uri")) {
+ pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->local.info->uri, buf, sizeof(buflen));
+ buf_copy = ast_strdupa(buf);
+ ast_escape_quoted(buf_copy, buf, buflen);
+ } else if (!strcmp(type, "remote_uri")) {
+ pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->remote.info->uri, buf, sizeof(buflen));
+ buf_copy = ast_strdupa(buf);
+ ast_escape_quoted(buf_copy, buf, buflen);
+ } else if (!strcmp(type, "t38state")) {
+ ast_copy_string(buf, t38state_to_string[channel->session->t38state], buflen);
+ } else if (!strcmp(type, "local_addr")) {
+ RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
+ struct transport_info_data *transport_data;
+
+ datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
+ if (!datastore) {
+ ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
+ return -1;
+ }
+ transport_data = datastore->data;
+
+ if (pj_sockaddr_has_addr(&transport_data->local_addr)) {
+ pj_sockaddr_print(&transport_data->local_addr, buf, buflen, 3);
+ }
+ } else if (!strcmp(type, "remote_addr")) {
+ RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
+ struct transport_info_data *transport_data;
+
+ datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
+ if (!datastore) {
+ ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
+ return -1;
+ }
+ transport_data = datastore->data;
+
+ if (pj_sockaddr_has_addr(&transport_data->remote_addr)) {
+ pj_sockaddr_print(&transport_data->remote_addr, buf, buflen, 3);
+ }
+ } else {
+ ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'pjsip' information\n", type);
+ return -1;
+ }
+
+ return 0;
+}
+
+/*! \brief Struct used to push function arguments to task processor */
+struct pjsip_func_args {
+ struct ast_channel *chan;
+ const char *param;
+ const char *type;
+ const char *field;
+ char *buf;
+ size_t len;
+ int ret;
+};
+
+/*! \internal \brief Taskprocessor callback that handles the read on a PJSIP thread */
+static int read_pjsip(void *data)
+{
+ struct pjsip_func_args *func_args = data;
+
+ if (!strcmp(func_args->param, "rtp")) {
+ func_args->ret = channel_read_rtp(func_args->chan, func_args->type,
+ func_args->field, func_args->buf,
+ func_args->len);
+ } else if (!strcmp(func_args->param, "rtcp")) {
+ func_args->ret = channel_read_rtcp(func_args->chan, func_args->type,
+ func_args->field, func_args->buf,
+ func_args->len);
+ } else if (!strcmp(func_args->param, "endpoint")) {
+ struct ast_sip_channel_pvt *pvt = ast_channel_tech_pvt(func_args->chan);
+
+ if (!pvt) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(func_args->chan));
+ return -1;
+ }
+ if (!pvt->session || !pvt->session->endpoint) {
+ ast_log(AST_LOG_WARNING, "Channel %s has no endpoint!\n", ast_channel_name(func_args->chan));
+ return -1;
+ }
+ snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(pvt->session->endpoint));
+ } else if (!strcmp(func_args->param, "pjsip")) {
+ func_args->ret = channel_read_pjsip(func_args->chan, func_args->type,
[... 594 lines stripped ...]
More information about the asterisk-commits
mailing list