[asterisk-commits] mmichelson: branch mmichelson/sip_endpoint_reorg r395746 - in /team/mmichelso...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 30 09:37:50 CDT 2013
Author: mmichelson
Date: Tue Jul 30 09:37:49 2013
New Revision: 395746
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=395746
Log:
Resolve conflict and reset automerge.
Added:
team/mmichelson/sip_endpoint_reorg/res/res_sip_t38.c
- copied, changed from r395731, trunk/res/res_sip_t38.c
Modified:
team/mmichelson/sip_endpoint_reorg/ (props changed)
team/mmichelson/sip_endpoint_reorg/channels/chan_gulp.c
team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip.h
team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip_session.h
team/mmichelson/sip_endpoint_reorg/res/res_pktccops.c
team/mmichelson/sip_endpoint_reorg/res/res_sip.c
team/mmichelson/sip_endpoint_reorg/res/res_sip/sip_configuration.c
team/mmichelson/sip_endpoint_reorg/res/res_sip_session.c
team/mmichelson/sip_endpoint_reorg/res/res_sip_session.exports.in
Propchange: team/mmichelson/sip_endpoint_reorg/
------------------------------------------------------------------------------
automerge = *
Propchange: team/mmichelson/sip_endpoint_reorg/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jul 30 09:37:49 2013
@@ -1,1 +1,1 @@
-/trunk:1-395690
+/trunk:1-395744
Modified: team/mmichelson/sip_endpoint_reorg/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/channels/chan_gulp.c?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/channels/chan_gulp.c (original)
+++ team/mmichelson/sip_endpoint_reorg/channels/chan_gulp.c Tue Jul 30 09:37:49 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 +433,8 @@
{
RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
- return ast_sip_session_refresh(session, NULL, NULL, session->endpoint->media.direct_media.method, 1);
+ return ast_sip_session_refresh(session, NULL, NULL, NULL,
+ session->endpoint->media.direct_media.method, 1);
}
static struct ast_datastore_info direct_media_mitigation_info = { };
@@ -668,6 +671,55 @@
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 +770,13 @@
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 +817,8 @@
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 +931,45 @@
}
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 +1092,7 @@
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 +1195,18 @@
} 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: team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip.h?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip.h (original)
+++ team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip.h Tue Jul 30 09:37:49 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 */
@@ -553,6 +555,18 @@
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;
};
Modified: team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip_session.h?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip_session.h (original)
+++ team/mmichelson/sip_endpoint_reorg/include/asterisk/res_sip_session.h Tue Jul 30 09:37:49 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: team/mmichelson/sip_endpoint_reorg/res/res_pktccops.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/res/res_pktccops.c?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/res/res_pktccops.c (original)
+++ team/mmichelson/sip_endpoint_reorg/res/res_pktccops.c Tue Jul 30 09:37:49 2013
@@ -1347,7 +1347,7 @@
if (a->argc < 9)
return CLI_SHOWUSAGE;
- if (!strncmp(a->argv[2], "null", sizeof(a->argv[2]))) {
+ if (!strcmp(a->argv[2], "null")) {
cmts = NULL;
} else {
AST_LIST_LOCK(&cmts_list);
Modified: team/mmichelson/sip_endpoint_reorg/res/res_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/res/res_sip.c?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/res/res_sip.c (original)
+++ team/mmichelson/sip_endpoint_reorg/res/res_sip.c Tue Jul 30 09:37:49 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: team/mmichelson/sip_endpoint_reorg/res/res_sip/sip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/res/res_sip/sip_configuration.c?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/res/res_sip/sip_configuration.c (original)
+++ team/mmichelson/sip_endpoint_reorg/res/res_sip/sip_configuration.c Tue Jul 30 09:37:49 2013
@@ -517,6 +517,24 @@
struct ast_sip_endpoint *endpoint = obj;
return ast_rtp_dtls_cfg_parse(&endpoint->media.rtp.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)
@@ -659,7 +677,12 @@
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", "rtpengine", "asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.rtp.engine));
+ 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", "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));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "recordonfeature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, info.recording.onfeature));
Modified: team/mmichelson/sip_endpoint_reorg/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/res/res_sip_session.c?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/res/res_sip_session.c (original)
+++ team/mmichelson/sip_endpoint_reorg/res/res_sip_session.c Tue Jul 30 09:37:49 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->extensions.timer.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: team/mmichelson/sip_endpoint_reorg/res/res_sip_session.exports.in
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/res/res_sip_session.exports.in?view=diff&rev=395746&r1=395745&r2=395746
==============================================================================
--- team/mmichelson/sip_endpoint_reorg/res/res_sip_session.exports.in (original)
+++ team/mmichelson/sip_endpoint_reorg/res/res_sip_session.exports.in Tue Jul 30 09:37:49 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:
*;
Copied: team/mmichelson/sip_endpoint_reorg/res/res_sip_t38.c (from r395731, trunk/res/res_sip_t38.c)
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/sip_endpoint_reorg/res/res_sip_t38.c?view=diff&rev=395746&p1=trunk/res/res_sip_t38.c&r1=395731&p2=team/mmichelson/sip_endpoint_reorg/res/res_sip_t38.c&r2=395746
==============================================================================
--- trunk/res/res_sip_t38.c (original)
+++ team/mmichelson/sip_endpoint_reorg/res/res_sip_t38.c Tue Jul 30 09:37:49 2013
@@ -671,7 +671,7 @@
media->desc.media = pj_str(session_media->stream_type);
media->desc.transport = STR_UDPTL;
- if (ast_strlen_zero(session->endpoint->external_media_address)) {
+ if (ast_strlen_zero(session->endpoint->media.external_address)) {
pj_sockaddr localaddr;
if (pj_gethostip(session->endpoint->t38udptl_ipv6 ? pj_AF_INET6() : pj_AF_INET(), &localaddr)) {
@@ -679,7 +679,7 @@
}
pj_sockaddr_print(&localaddr, hostip, sizeof(hostip), 2);
} else {
- ast_copy_string(hostip, session->endpoint->external_media_address, sizeof(hostip));
+ ast_copy_string(hostip, session->endpoint->media.external_address, sizeof(hostip));
}
media->conn->net_type = STR_IN;
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