[asterisk-commits] file: trunk r395731 - in /trunk: channels/ include/asterisk/ res/ res/res_sip/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 30 09:16:44 CDT 2013
Author: file
Date: Tue Jul 30 09:16:41 2013
New Revision: 395731
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=395731
Log:
Add support for T.38 fax to chan_pjsip.
Review: https://reviewboard.asterisk.org/r/2692/
Added:
trunk/res/res_sip_t38.c (with props)
Modified:
trunk/channels/chan_gulp.c
trunk/include/asterisk/res_sip.h
trunk/include/asterisk/res_sip_session.h
trunk/res/res_sip.c
trunk/res/res_sip/sip_configuration.c
trunk/res/res_sip_session.c
trunk/res/res_sip_session.exports.in
Modified: trunk/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_gulp.c?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/channels/chan_gulp.c (original)
+++ trunk/channels/chan_gulp.c Tue Jul 30 09:16:41 2013
@@ -142,6 +142,7 @@
static int gulp_transfer(struct ast_channel *ast, const char *target);
static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int gulp_devicestate(const char *data);
+static int gulp_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
/*! \brief PBX interface structure for channel registration */
static struct ast_channel_tech gulp_tech = {
@@ -162,6 +163,7 @@
.transfer = gulp_transfer,
.fixup = gulp_fixup,
.devicestate = gulp_devicestate,
+ .queryoption = gulp_queryoption,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
@@ -431,7 +433,7 @@
{
RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
- return ast_sip_session_refresh(session, NULL, NULL, session->endpoint->direct_media_method, 1);
+ return ast_sip_session_refresh(session, NULL, NULL, NULL, session->endpoint->direct_media_method, 1);
}
static struct ast_datastore_info direct_media_mitigation_info = { };
@@ -668,6 +670,55 @@
return 0;
}
+/*! \brief Internal helper function called when CNG tone is detected */
+static struct ast_frame *gulp_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
+{
+ const char *target_context;
+ int exists;
+
+ /* If we only needed this DSP for fax detection purposes we can just drop it now */
+ if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+ ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
+ } else {
+ ast_dsp_free(session->dsp);
+ session->dsp = NULL;
+ }
+
+ /* If already executing in the fax extension don't do anything */
+ if (!strcmp(ast_channel_exten(session->channel), "fax")) {
+ return f;
+ }
+
+ target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
+
+ /* We need to unlock the channel here because ast_exists_extension has the
+ * potential to start and stop an autoservice on the channel. Such action
+ * is prone to deadlock if the channel is locked.
+ */
+ ast_channel_unlock(session->channel);
+ exists = ast_exists_extension(session->channel, target_context, "fax", 1,
+ S_COR(ast_channel_caller(session->channel)->id.number.valid,
+ ast_channel_caller(session->channel)->id.number.str, NULL));
+ ast_channel_lock(session->channel);
+
+ if (exists) {
+ ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
+ ast_channel_name(session->channel));
+ pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
+ if (ast_async_goto(session->channel, target_context, "fax", 1)) {
+ ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
+ ast_channel_name(session->channel), target_context);
+ }
+ ast_frfree(f);
+ f = &ast_null_frame;
+ } else {
+ ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
+ ast_channel_name(session->channel), target_context);
+ }
+
+ return f;
+}
+
/*! \brief Function called by core to read any waiting frames */
static struct ast_frame *gulp_read(struct ast_channel *ast)
{
@@ -718,8 +769,13 @@
f = ast_dsp_process(ast, channel->session->dsp, f);
if (f && (f->frametype == AST_FRAME_DTMF)) {
- ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
- ast_channel_name(ast));
+ if (f->subclass.integer == 'f') {
+ ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
+ f = gulp_cng_tone_detected(channel->session, f);
+ } else {
+ ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
+ ast_channel_name(ast));
+ }
}
}
@@ -760,6 +816,8 @@
if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
res = ast_rtp_instance_write(media->rtp, frame);
}
+ break;
+ case AST_FRAME_MODEM:
break;
default:
ast_log(LOG_WARNING, "Can't send %d type frames with Gulp\n", frame->frametype);
@@ -872,6 +930,45 @@
}
return state;
+}
+
+/*! \brief Function called to query options on a channel */
+static int gulp_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
+{
+ struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = channel->session;
+ int res = -1;
+ enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
+
+ switch (option) {
+ case AST_OPTION_T38_STATE:
+ if (session->endpoint->t38udptl) {
+ switch (session->t38state) {
+ case T38_LOCAL_REINVITE:
+ case T38_PEER_REINVITE:
+ state = T38_STATE_NEGOTIATING;
+ break;
+ case T38_ENABLED:
+ state = T38_STATE_NEGOTIATED;
+ break;
+ case T38_REJECTED:
+ state = T38_STATE_REJECTED;
+ break;
+ default:
+ state = T38_STATE_UNKNOWN;
+ break;
+ }
+ }
+
+ *((enum ast_t38_state *) data) = state;
+ res = 0;
+
+ break;
+ default:
+ break;
+ }
+
+ return res;
}
struct indicate_data {
@@ -994,7 +1091,7 @@
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
}
- ast_sip_session_refresh(session, NULL, NULL, method, 0);
+ ast_sip_session_refresh(session, NULL, NULL, NULL, method, 0);
}
return 0;
@@ -1097,6 +1194,18 @@
} else {
res = -1;
}
+ break;
+ case AST_CONTROL_T38_PARAMETERS:
+ res = 0;
+
+ if (channel->session->t38state == T38_PEER_REINVITE) {
+ const struct ast_control_t38_parameters *parameters = data;
+
+ if (parameters->request_response == AST_T38_REQUEST_PARMS) {
+ res = AST_T38_REQUEST_PARMS;
+ }
+ }
+
break;
case -1:
res = -1;
Modified: trunk/include/asterisk/res_sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_sip.h?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/include/asterisk/res_sip.h (original)
+++ trunk/include/asterisk/res_sip.h Tue Jul 30 09:16:41 2013
@@ -32,6 +32,8 @@
#include "asterisk/dnsmgr.h"
/* Needed for ast_endpoint */
#include "asterisk/endpoints.h"
+/* Needed for ast_t38_ec_modes */
+#include "asterisk/udptl.h"
/* Needed for pj_sockaddr */
#include <pjlib.h>
/* Needed for ast_rtp_dtls_cfg struct */
@@ -434,6 +436,18 @@
struct ast_endpoint *persistent;
/*! The number of channels at which busy device state is returned */
unsigned int devicestate_busy_at;
+ /*! Whether T.38 UDPTL support is enabled or not */
+ unsigned int t38udptl;
+ /*! Error correction setting for T.38 UDPTL */
+ enum ast_t38_ec_modes t38udptl_ec;
+ /*! Explicit T.38 max datagram value, may be 0 to indicate the remote side can be trusted */
+ unsigned int t38udptl_maxdatagram;
+ /*! Whether fax detection is enabled or not (CNG tone detection) */
+ unsigned int faxdetect;
+ /*! Whether NAT Support is enabled for T.38 UDPTL sessions or not */
+ unsigned int t38udptl_nat;
+ /*! Whether to use IPv6 for UDPTL or not */
+ unsigned int t38udptl_ipv6;
/*! Determines if transfers (using REFER) are allowed by this endpoint */
unsigned int allowtransfer;
/*! DSCP TOS bits for audio streams */
Modified: trunk/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_sip_session.h?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/include/asterisk/res_sip_session.h (original)
+++ trunk/include/asterisk/res_sip_session.h Tue Jul 30 09:16:41 2013
@@ -43,6 +43,16 @@
struct pjmedia_sdp_media;
struct pjmedia_sdp_session;
struct ast_dsp;
+struct ast_udptl;
+
+/*! \brief T.38 states for a session */
+enum ast_sip_session_t38state {
+ T38_DISABLED = 0, /*!< Not enabled */
+ T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
+ T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
+ T38_ENABLED, /*!< Negotiated (enabled) */
+ T38_REJECTED, /*!< Refused */
+};
struct ast_sip_session_sdp_handler;
@@ -50,8 +60,12 @@
* \brief A structure containing SIP session media information
*/
struct ast_sip_session_media {
- /*! \brief RTP instance itself */
- struct ast_rtp_instance *rtp;
+ union {
+ /*! \brief RTP instance itself */
+ struct ast_rtp_instance *rtp;
+ /*! \brief UDPTL instance itself */
+ struct ast_udptl *udptl;
+ };
/*! \brief Direct media address */
struct ast_sockaddr direct_media_addr;
/*! \brief SDP handler that setup the RTP */
@@ -113,10 +127,15 @@
struct ast_dsp *dsp;
/* Whether the termination of the session should be deferred */
unsigned int defer_terminate:1;
+ /* Deferred incoming re-invite */
+ pjsip_rx_data *deferred_reinvite;
+ /* Current T.38 state */
+ enum ast_sip_session_t38state t38state;
};
typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);
typedef int (*ast_sip_session_response_cb)(struct ast_sip_session *session, pjsip_rx_data *rdata);
+typedef int (*ast_sip_session_sdp_creation_cb)(struct ast_sip_session *session, pjmedia_sdp_session *sdp);
enum ast_sip_session_supplement_priority {
/*! Top priority. Supplements with this priority are those that need to run before any others */
@@ -210,6 +229,19 @@
/*! An identifier for this handler */
const char *id;
/*!
+ * \brief Determine whether a stream requires that the re-invite be deferred.
+ * If a stream can not be immediately negotiated the re-invite can be deferred and
+ * resumed at a later time. It is up to the handler which caused deferral to occur
+ * to resume it.
+ * \param session The session for which the media is being re-invited
+ * \param session_media The media being reinvited
+ * \param sdp The entire SDP.
+ * \retval 0 The stream was unhandled or does not need the re-invite to be deferred.
+ * \retval 1 Re-invite should be deferred and will be resumed later. No further operations will take place.
+ * \note This is optional, if not implemented the stream is assumed to not be deferred.
+ */
+ int (*defer_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream);
+ /*!
* \brief Set session details based on a stream in an incoming SDP offer or answer
* \param session The session for which the media is being negotiated
* \param session_media The media to be setup for this session
@@ -443,6 +475,7 @@
*
* \param session The session on which the reinvite will be sent
* \param on_request_creation Callback called when request is created
+ * \param on_sdp_creation Callback called when SDP is created
* \param on_response Callback called when response for request is received
* \param method The method that should be used when constructing the session refresh
* \param generate_new_sdp Boolean to indicate if a new SDP should be created
@@ -451,6 +484,7 @@
*/
int ast_sip_session_refresh(struct ast_sip_session *session,
ast_sip_session_request_creation_cb on_request_creation,
+ ast_sip_session_sdp_creation_cb on_sdp_creation,
ast_sip_session_response_cb on_response,
enum ast_sip_session_refresh_method method,
int generate_new_sdp);
@@ -513,4 +547,15 @@
*/
struct ast_sip_session *ast_sip_dialog_get_session(pjsip_dialog *dlg);
+/*!
+ * \brief Resumes processing of a deferred incoming re-invite
+ *
+ * \param session The session which has a pending incoming re-invite
+ *
+ * \note When resuming a re-invite it is given to the pjsip stack as if it
+ * had just been received from a transport, this means that the deferral
+ * callback will be called again.
+ */
+void ast_sip_session_resume_reinvite(struct ast_sip_session *session);
+
#endif /* _RES_SIP_SESSION_H */
Modified: trunk/res/res_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip.c?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/res/res_sip.c (original)
+++ trunk/res/res_sip.c Tue Jul 30 09:16:41 2013
@@ -398,6 +398,56 @@
Gulp channel driver will return busy as the device state instead of in use.
</para></description>
</configOption>
+ <configOption name="t38udptl" default="no">
+ <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
+ <description><para>
+ If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
+ and relayed.
+ </para></description>
+ </configOption>
+ <configOption name="t38udptl_ec" default="none">
+ <synopsis>T.38 UDPTL error correction method</synopsis>
+ <description>
+ <enumlist>
+ <enum name="none"><para>
+ No error correction should be used.
+ </para></enum>
+ <enum name="fec"><para>
+ Forward error correction should be used.
+ </para></enum>
+ <enum name="redundancy"><para>
+ Redundacy error correction should be used.
+ </para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="t38udptl_maxdatagram" default="0">
+ <synopsis>T.38 UDPTL maximum datagram size</synopsis>
+ <description><para>
+ This option can be set to override the maximum datagram of a remote endpoint for broken
+ endpoints.
+ </para></description>
+ </configOption>
+ <configOption name="faxdetect" default="no">
+ <synopsis>Whether CNG tone detection is enabled</synopsis>
+ <description><para>
+ This option can be set to send the session to the fax extension when a CNG tone is
+ detected.
+ </para></description>
+ </configOption>
+ <configOption name="t38udptl_nat" default="no">
+ <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
+ <description><para>
+ When enabled the UDPTL stack will send UDPTL packets to the source address of
+ received packets.
+ </para></description>
+ </configOption>
+ <configOption name="t38udptl_ipv6" default="no">
+ <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
+ <description><para>
+ When enabled the UDPTL stack will use IPv6.
+ </para></description>
+ </configOption>
<configOption name="tonezone">
<synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
</configOption>
Modified: trunk/res/res_sip/sip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip/sip_configuration.c?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/res/res_sip/sip_configuration.c (original)
+++ trunk/res/res_sip/sip_configuration.c Tue Jul 30 09:16:41 2013
@@ -525,6 +525,24 @@
struct ast_sip_endpoint *endpoint = obj;
return ast_rtp_dtls_cfg_parse(&endpoint->dtls_cfg, var->name, var->value);
+}
+
+static int t38udptl_ec_handler(const struct aco_option *opt,
+ struct ast_variable *var, void *obj)
+{
+ struct ast_sip_endpoint *endpoint = obj;
+
+ if (!strcmp(var->value, "none")) {
+ endpoint->t38udptl_ec = UDPTL_ERROR_CORRECTION_NONE;
+ } else if (!strcmp(var->value, "fec")) {
+ endpoint->t38udptl_ec = UDPTL_ERROR_CORRECTION_FEC;
+ } else if (!strcmp(var->value, "redundancy")) {
+ endpoint->t38udptl_ec = UDPTL_ERROR_CORRECTION_REDUNDANCY;
+ } else {
+ return -1;
+ }
+
+ return 0;
}
static void *sip_nat_hook_alloc(const char *name)
@@ -667,6 +685,12 @@
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "namedcallgroup", "", named_groups_handler, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "namedpickupgroup", "", named_groups_handler, NULL, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "devicestate_busy_at", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, devicestate_busy_at));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "t38udptl", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, t38udptl));
+ ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "t38udptl_ec", "none", t38udptl_ec_handler, NULL, 0, 0);
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "t38udptl_maxdatagram", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, t38udptl_maxdatagram));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "faxdetect", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, faxdetect));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "t38udptl_nat", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, t38udptl_nat));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "t38udptl_ipv6", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, t38udptl_ipv6));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtpengine", "asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, rtp_engine));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "tonezone", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, zone));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "language", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, language));
Modified: trunk/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_session.c?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/res/res_sip_session.c (original)
+++ trunk/res/res_sip_session.c Tue Jul 30 09:16:41 2013
@@ -412,6 +412,10 @@
struct ast_sip_session_sdp_handler *handler;
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
+ if (!remote->media[i]) {
+ continue;
+ }
+
/* We need a null-terminated version of the media string */
ast_copy_pj_str(media, &local->media[i]->desc.media, sizeof(media));
@@ -602,6 +606,8 @@
char method[15];
/*! Callback to call when the delayed request is created. */
ast_sip_session_request_creation_cb on_request_creation;
+ /*! Callback to call when the delayed request SDP is created */
+ ast_sip_session_sdp_creation_cb on_sdp_creation;
/*! Callback to call when the delayed request receives a response */
ast_sip_session_response_cb on_response;
/*! Request to send */
@@ -611,6 +617,7 @@
static struct ast_sip_session_delayed_request *delayed_request_alloc(const char *method,
ast_sip_session_request_creation_cb on_request_creation,
+ ast_sip_session_sdp_creation_cb on_sdp_creation,
ast_sip_session_response_cb on_response,
pjsip_tx_data *tdata)
{
@@ -620,6 +627,7 @@
}
ast_copy_string(delay->method, method, sizeof(delay->method));
delay->on_request_creation = on_request_creation;
+ delay->on_sdp_creation = on_sdp_creation;
delay->on_response = on_response;
delay->tdata = tdata;
return delay;
@@ -636,10 +644,10 @@
if (!strcmp(delay->method, "INVITE")) {
ast_sip_session_refresh(session, delay->on_request_creation,
- delay->on_response, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1);
+ delay->on_sdp_creation, delay->on_response, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1);
} else if (!strcmp(delay->method, "UPDATE")) {
ast_sip_session_refresh(session, delay->on_request_creation,
- delay->on_response, AST_SIP_SESSION_REFRESH_METHOD_UPDATE, 1);
+ delay->on_sdp_creation, delay->on_response, AST_SIP_SESSION_REFRESH_METHOD_UPDATE, 1);
} else {
ast_log(LOG_WARNING, "Unexpected delayed %s request with no existing request structure\n", delay->method);
return -1;
@@ -675,10 +683,11 @@
}
static int delay_request(struct ast_sip_session *session, ast_sip_session_request_creation_cb on_request,
- ast_sip_session_response_cb on_response, const char *method, pjsip_tx_data *tdata)
+ ast_sip_session_sdp_creation_cb on_sdp_creation, ast_sip_session_response_cb on_response,
+ const char *method, pjsip_tx_data *tdata)
{
struct ast_sip_session_delayed_request *delay = delayed_request_alloc(method,
- on_request, on_response, tdata);
+ on_request, on_sdp_creation, on_response, tdata);
if (!delay) {
return -1;
@@ -702,7 +711,9 @@
}
int ast_sip_session_refresh(struct ast_sip_session *session,
- ast_sip_session_request_creation_cb on_request_creation, ast_sip_session_response_cb on_response,
+ ast_sip_session_request_creation_cb on_request_creation,
+ ast_sip_session_sdp_creation_cb on_sdp_creation,
+ ast_sip_session_response_cb on_response,
enum ast_sip_session_refresh_method method, int generate_new_sdp)
{
pjsip_inv_session *inv_session = session->inv_session;
@@ -721,7 +732,7 @@
/* We can't send a reinvite yet, so delay it */
ast_debug(3, "Delaying sending reinvite to %s because of outstanding transaction...\n",
ast_sorcery_object_get_id(session->endpoint));
- return delay_request(session, on_request_creation, on_response, "INVITE", NULL);
+ return delay_request(session, on_request_creation, on_sdp_creation, on_response, "INVITE", NULL);
} else if (inv_session->state != PJSIP_INV_STATE_CONFIRMED) {
/* Initial INVITE transaction failed to progress us to a confirmed state
* which means re-invites are not possible
@@ -738,6 +749,11 @@
ast_log(LOG_ERROR, "Failed to generate session refresh SDP. Not sending session refresh\n");
return -1;
}
+ if (on_sdp_creation) {
+ if (on_sdp_creation(session, new_sdp)) {
+ return -1;
+ }
+ }
}
if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
@@ -771,6 +787,132 @@
.name = {"Session Module", 14},
.priority = PJSIP_MOD_PRIORITY_APPLICATION,
.on_rx_request = session_on_rx_request,
+};
+
+/*! \brief Determine whether the SDP provided requires deferral of negotiating or not
+ *
+ * \retval 1 re-invite should be deferred and resumed later
+ * \retval 0 re-invite should not be deferred
+ */
+static int sdp_requires_deferral(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
+{
+ int i;
+ if (validate_incoming_sdp(sdp)) {
+ return 0;
+ }
+
+ for (i = 0; i < sdp->media_count; ++i) {
+ /* See if there are registered handlers for this media stream type */
+ char media[20];
+ struct ast_sip_session_sdp_handler *handler;
+ RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
+ RAII_VAR(struct ast_sip_session_media *, session_media, NULL, ao2_cleanup);
+
+ /* We need a null-terminated version of the media string */
+ ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));
+
+ session_media = ao2_find(session->media, media, OBJ_KEY);
+ if (!session_media) {
+ /* if the session_media doesn't exist, there weren't
+ * any handlers at the time of its creation */
+ continue;
+ }
+
+ if (session_media->handler && session_media->handler->defer_incoming_sdp_stream) {
+ int res;
+ handler = session_media->handler;
+ res = handler->defer_incoming_sdp_stream(
+ session, session_media, sdp, sdp->media[i]);
+ if (res) {
+ return 1;
+ }
+ }
+
+ handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
+ if (!handler_list) {
+ ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
+ continue;
+ }
+ AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
+ int res;
+ if (session_media->handler) {
+ /* There is only one slot for this stream type and it has already been claimed
+ * so it will go unhandled */
+ break;
+ }
+ if (!handler->defer_incoming_sdp_stream) {
+ continue;
+ }
+ res = handler->defer_incoming_sdp_stream(session, session_media, sdp, sdp->media[i]);
+ if (res) {
+ return 1;
+ }
+ }
+ }
+ return 0;
+}
+
+static pj_bool_t session_reinvite_on_rx_request(pjsip_rx_data *rdata)
+{
+ pjsip_dialog *dlg;
+ RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
+ pjsip_rdata_sdp_info *sdp_info;
+
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD ||
+ !(dlg = pjsip_ua_find_dialog(&rdata->msg_info.cid->id, &rdata->msg_info.to->tag, &rdata->msg_info.from->tag, PJ_FALSE)) ||
+ !(session = ast_sip_dialog_get_session(dlg))) {
+ return PJ_FALSE;
+ }
+
+ if (session->deferred_reinvite) {
+ pj_str_t key, deferred_key;
+ pjsip_tx_data *tdata;
+
+ /* We use memory from the new request on purpose so the deferred reinvite pool does not grow uncontrollably */
+ pjsip_tsx_create_key(rdata->tp_info.pool, &key, PJSIP_ROLE_UAS, &rdata->msg_info.cseq->method, rdata);
+ pjsip_tsx_create_key(rdata->tp_info.pool, &deferred_key, PJSIP_ROLE_UAS, &session->deferred_reinvite->msg_info.cseq->method,
+ session->deferred_reinvite);
+
+ /* If this is a retransmission ignore it */
+ if (!pj_strcmp(&key, &deferred_key)) {
+ return PJ_TRUE;
+ }
+
+ /* Otherwise this is a new re-invite, so reject it */
+ if (pjsip_dlg_create_response(dlg, rdata, 491, NULL, &tdata) == PJ_SUCCESS) {
+ pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL);
+ }
+
+ return PJ_TRUE;
+ }
+
+ if (!(sdp_info = pjsip_rdata_get_sdp_info(rdata)) ||
+ (sdp_info->sdp_err != PJ_SUCCESS) ||
+ !sdp_info->sdp ||
+ !sdp_requires_deferral(session, sdp_info->sdp)) {
+ return PJ_FALSE;
+ }
+
+ pjsip_rx_data_clone(rdata, 0, &session->deferred_reinvite);
+
+ return PJ_TRUE;
+}
+
+void ast_sip_session_resume_reinvite(struct ast_sip_session *session)
+{
+ if (!session->deferred_reinvite) {
+ return;
+ }
+
+ pjsip_endpt_process_rx_data(ast_sip_get_pjsip_endpoint(), session->deferred_reinvite, NULL, NULL);
+ pjsip_rx_data_free_cloned(session->deferred_reinvite);
+ session->deferred_reinvite = NULL;
+}
+
+static pjsip_module session_reinvite_module = {
+ .name = { "Session Re-Invite Module", 24 },
+ .priority = PJSIP_MOD_PRIORITY_UA_PROXY_LAYER - 1,
+ .on_rx_request = session_reinvite_on_rx_request,
};
void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata,
@@ -940,6 +1082,7 @@
{
RAII_VAR(struct ast_sip_session *, session, ao2_alloc(sizeof(*session), session_destructor), ao2_cleanup);
struct ast_sip_session_supplement *iter;
+ int dsp_features = 0;
if (!session) {
return NULL;
}
@@ -971,12 +1114,20 @@
session->req_caps = ast_format_cap_alloc_nolock();
if (endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+ dsp_features |= DSP_FEATURE_DIGIT_DETECT;
+ }
+
+ if (endpoint->faxdetect) {
+ dsp_features |= DSP_FEATURE_FAX_DETECT;
+ }
+
+ if (dsp_features) {
if (!(session->dsp = ast_dsp_new())) {
ao2_ref(session, -1);
return NULL;
}
- ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
+ ast_dsp_set_features(session->dsp, dsp_features);
}
if (add_supplements(session)) {
@@ -1044,6 +1195,9 @@
pjsip_dlg_terminate(dlg);
return NULL;
}
+#ifdef PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE
+ inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
+#endif
pjsip_timer_setting_default(&timer);
timer.min_se = endpoint->min_se;
@@ -1189,6 +1343,9 @@
pjsip_dlg_terminate(dlg);
return NULL;
}
+#ifdef PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE
+ inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
+#endif
if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
@@ -1473,7 +1630,7 @@
static void reschedule_reinvite(struct ast_sip_session *session, ast_sip_session_response_cb on_response, pjsip_tx_data *tdata)
{
struct ast_sip_session_delayed_request *delay = delayed_request_alloc("INVITE",
- NULL, on_response, tdata);
+ NULL, NULL, on_response, tdata);
pjsip_inv_session *inv = session->inv_session;
struct reschedule_reinvite_data *rrd = reschedule_reinvite_data_alloc(session, delay);
pj_time_val tv;
@@ -1710,8 +1867,9 @@
if (tsx->status_code == PJSIP_SC_REQUEST_PENDING) {
reschedule_reinvite(session, tsx->mod_data[session_module.id], tsx->last_tx);
return;
- } else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
- /* Other reinvite failures result in destroying the session. */
+ } else if (inv->state == PJSIP_INV_STATE_CONFIRMED &&
+ tsx->status_code != 488) {
+ /* Other reinvite failures (except 488) result in destroying the session. */
pjsip_tx_data *tdata;
if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS) {
ast_sip_session_send_request(session, tdata);
@@ -1952,12 +2110,14 @@
if (ast_sip_register_service(&session_module)) {
return AST_MODULE_LOAD_DECLINE;
}
+ ast_sip_register_service(&session_reinvite_module);
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_unregister_service(&session_module);
+ ast_sip_unregister_service(&session_reinvite_module);
if (nat_hook) {
ast_sorcery_delete(ast_sip_get_sorcery(), nat_hook);
nat_hook = NULL;
Modified: trunk/res/res_sip_session.exports.in
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_session.exports.in?view=diff&rev=395731&r1=395730&r2=395731
==============================================================================
--- trunk/res/res_sip_session.exports.in (original)
+++ trunk/res/res_sip_session.exports.in Tue Jul 30 09:16:41 2013
@@ -16,6 +16,7 @@
LINKER_SYMBOL_PREFIXast_sip_session_create_invite;
LINKER_SYMBOL_PREFIXast_sip_session_create_outgoing;
LINKER_SYMBOL_PREFIXast_sip_dialog_get_session;
+ LINKER_SYMBOL_PREFIXast_sip_session_resume_reinvite;
LINKER_SYMBOL_PREFIXast_sip_channel_pvt_alloc;
local:
*;
Added: trunk/res/res_sip_t38.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_t38.c?view=auto&rev=395731
==============================================================================
--- trunk/res/res_sip_t38.c (added)
+++ trunk/res/res_sip_t38.c Tue Jul 30 09:16:41 2013
@@ -1,0 +1,853 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, 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 SIP T.38 handling
+ */
+
+/*** MODULEINFO
+ <depend>pjproject</depend>
+ <depend>res_sip</depend>
+ <depend>res_sip_session</depend>
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+#include <pjmedia.h>
+#include <pjlib.h>
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/module.h"
+#include "asterisk/udptl.h"
+#include "asterisk/netsock2.h"
+#include "asterisk/channel.h"
+#include "asterisk/acl.h"
+
+#include "asterisk/res_sip.h"
+#include "asterisk/res_sip_session.h"
+
+/*! \brief The number of seconds after receiving a T.38 re-invite before automatically rejecting it */
+#define T38_AUTOMATIC_REJECTION_SECONDS 5
+
+/*! \brief Address for IPv4 UDPTL */
+static struct ast_sockaddr address_ipv4;
+
+/*! \brief Address for IPv6 UDPTL */
+static struct ast_sockaddr address_ipv6;
+
+/*! \brief T.38 state information */
+struct t38_state {
+ /*! \brief Current state */
+ enum ast_sip_session_t38state state;
+ /*! \brief Our T.38 parameters */
+ struct ast_control_t38_parameters our_parms;
+ /*! \brief Their T.38 parameters */
+ struct ast_control_t38_parameters their_parms;
+ /*! \brief Timer entry for automatically rejecting an inbound re-invite */
+ pj_timer_entry timer;
+};
+
+/*! \brief Destructor for T.38 state information */
+static void t38_state_destroy(void *obj)
+{
+ ast_free(obj);
+}
+
+/*! \brief Datastore for attaching T.38 state information */
+static const struct ast_datastore_info t38_datastore = {
+ .type = "t38",
+ .destroy = t38_state_destroy,
+};
+
+/*! \brief Structure for T.38 parameters task data */
+struct t38_parameters_task_data {
+ /*! \brief Session itself */
+ struct ast_sip_session *session;
+ /*! \brief T.38 control frame */
+ struct ast_frame *frame;
+};
+
+/*! \brief Destructor for T.38 data */
+static void t38_parameters_task_data_destroy(void *obj)
+{
+ struct t38_parameters_task_data *data = obj;
+
+ ao2_cleanup(data->session);
+
+ if (data->frame) {
+ ast_frfree(data->frame);
+ }
+}
+
+/*! \brief Allocator for T.38 data */
+static struct t38_parameters_task_data *t38_parameters_task_data_alloc(struct ast_sip_session *session,
+ struct ast_frame *frame)
+{
+ struct t38_parameters_task_data *data = ao2_alloc(sizeof(*data), t38_parameters_task_data_destroy);
+
+ if (!data) {
+ return NULL;
+ }
+
+ data->session = session;
+ ao2_ref(session, +1);
+
+ if (!(data->frame = ast_frdup(frame))) {
+ ao2_cleanup(data);
+ return NULL;
+ }
+
+ return data;
+}
+
+/*! \brief Helper function for changing the T.38 state */
+static void t38_change_state(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
+ struct t38_state *state, enum ast_sip_session_t38state new_state)
+{
+ enum ast_sip_session_t38state old_state = session->t38state;
+ struct ast_control_t38_parameters parameters = { .request_response = 0, };
+ pj_time_val delay = { .sec = T38_AUTOMATIC_REJECTION_SECONDS };
+
+ if (old_state == new_state) {
+ return;
+ }
+
+ session->t38state = new_state;
+ ast_debug(2, "T.38 state changed to '%d' from '%d' on channel '%s'\n", new_state, old_state, ast_channel_name(session->channel));
+
+ if (pj_timer_heap_cancel(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()), &state->timer)) {
+ ast_debug(2, "Automatic T.38 rejection on channel '%s' terminated\n", ast_channel_name(session->channel));
+ ao2_ref(session, -1);
+ }
+
+ if (!session->channel) {
+ return;
+ }
+
+ switch (new_state) {
+ case T38_PEER_REINVITE:
+ ao2_ref(session, +1);
+ if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(), &state->timer, &delay) != PJ_SUCCESS) {
+ ast_log(LOG_WARNING, "Scheduling of automatic T.38 rejection for channel '%s' failed\n",
+ ast_channel_name(session->channel));
+ ao2_ref(session, -1);
+ }
+ parameters = state->their_parms;
+ parameters.max_ifp = ast_udptl_get_far_max_ifp(session_media->udptl);
+ parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
+ ast_udptl_set_tag(session_media->udptl, "%s", ast_channel_name(session->channel));
+ break;
+ case T38_ENABLED:
+ parameters = state->their_parms;
+ parameters.max_ifp = ast_udptl_get_far_max_ifp(session_media->udptl);
+ parameters.request_response = AST_T38_NEGOTIATED;
+ ast_udptl_set_tag(session_media->udptl, "%s", ast_channel_name(session->channel));
+ break;
+ case T38_REJECTED:
+ case T38_DISABLED:
+ if (old_state == T38_ENABLED) {
+ parameters.request_response = AST_T38_TERMINATED;
+ } else if (old_state == T38_LOCAL_REINVITE) {
+ parameters.request_response = AST_T38_REFUSED;
+ }
+ break;
+ case T38_LOCAL_REINVITE:
+ /* wait until we get a peer response before responding to local reinvite */
+ break;
+ }
+
+ if (parameters.request_response) {
+ ast_queue_control_data(session->channel, AST_CONTROL_T38_PARAMETERS, ¶meters, sizeof(parameters));
+ }
+}
+
+/*! \brief Task function which rejects a T.38 re-invite and resumes handling it */
+static int t38_automatic_reject(void *obj)
+{
+ RAII_VAR(struct ast_sip_session *, session, obj, ao2_cleanup);
+ RAII_VAR(struct ast_datastore *, datastore, ast_sip_session_get_datastore(session, "t38"), ao2_cleanup);
+ RAII_VAR(struct ast_sip_session_media *, session_media, ao2_find(session->media, "image", OBJ_KEY), ao2_cleanup);
+
+ if (!datastore) {
+ return 0;
+ }
+
+ ast_debug(2, "Automatically rejecting T.38 request on channel '%s'\n", ast_channel_name(session->channel));
+
+ t38_change_state(session, session_media, datastore->data, T38_REJECTED);
+ ast_sip_session_resume_reinvite(session);
+
+ return 0;
+}
+
+/*! \brief Timer entry callback which queues a task to reject a T.38 re-invite and resume handling it */
+static void t38_automatic_reject_timer_cb(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry)
+{
+ struct ast_sip_session *session = entry->user_data;
+
+ if (ast_sip_push_task(session->serializer, t38_automatic_reject, session)) {
+ ao2_ref(session, -1);
+ }
+}
+
+/*! \brief Helper function which retrieves or allocates a T.38 state information datastore */
+static struct t38_state *t38_state_get_or_alloc(struct ast_sip_session *session)
+{
+ RAII_VAR(struct ast_datastore *, datastore, ast_sip_session_get_datastore(session, "t38"), ao2_cleanup);
+ struct t38_state *state;
+
+ /* While the datastore refcount is decremented this is operating in the serializer so it will remain valid regardless */
+ if (datastore) {
+ return datastore->data;
+ }
+
+ if (!(datastore = ast_sip_session_alloc_datastore(&t38_datastore, "t38")) ||
+ !(datastore->data = ast_calloc(1, sizeof(struct t38_state))) ||
+ ast_sip_session_add_datastore(session, datastore)) {
+ return NULL;
+ }
+
+ state = datastore->data;
+
+ /* This will get bumped up before scheduling */
+ state->timer.user_data = session;
+ state->timer.cb = t38_automatic_reject_timer_cb;
+
+ datastore->data = state;
+
+ return state;
+}
+
+/*! \brief Initializes UDPTL support on a session, only done when actually needed */
+static int t38_initialize_session(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
+{
+ if (session_media->udptl) {
+ return 0;
+ }
+
+ if (!(session_media->udptl = ast_udptl_new_with_bindaddr(NULL, NULL, 0,
+ session->endpoint->t38udptl_ipv6 ? &address_ipv6 : &address_ipv4))) {
+ return -1;
+ }
+
+ ast_channel_set_fd(session->channel, 5, ast_udptl_fd(session_media->udptl));
+ ast_udptl_set_error_correction_scheme(session_media->udptl, session->endpoint->t38udptl_ec);
+ ast_udptl_setnat(session_media->udptl, session->endpoint->t38udptl_nat);
+
+ return 0;
+}
+
+/*! \brief Callback for when T.38 reinvite SDP is created */
+static int t38_reinvite_sdp_cb(struct ast_sip_session *session, pjmedia_sdp_session *sdp)
+{
+ int stream;
+
+ /* Move the image media stream to the front and have it as the only stream, pjmedia will fill in
+ * dummy streams for the rest
+ */
+ for (stream = 0; stream < sdp->media_count++; ++stream) {
+ if (!pj_strcmp2(&sdp->media[stream]->desc.media, "image")) {
+ sdp->media[0] = sdp->media[stream];
+ sdp->media_count = 1;
+ break;
+ }
+ }
+
+ return 0;
+}
+
+/*! \brief Callback for when a response is received for a T.38 re-invite */
+static int t38_reinvite_response_cb(struct ast_sip_session *session, pjsip_rx_data *rdata)
+{
+ struct pjsip_status_line status = rdata->msg_info.msg->line.status;
+ struct t38_state *state;
+ RAII_VAR(struct ast_sip_session_media *, session_media, NULL, ao2_cleanup);
+
+ if (status.code == 100) {
+ return 0;
+ }
+
+ if (!(state = t38_state_get_or_alloc(session)) ||
+ !(session_media = ao2_find(session->media, "image", OBJ_KEY))) {
+ ast_log(LOG_WARNING, "Received response to T.38 re-invite on '%s' but state unavailable\n",
+ ast_channel_name(session->channel));
+ return 0;
+ }
+
[... 568 lines stripped ...]
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