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 <[email protected]>
+ *
+ * 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 <[email protected]>
+ *
+ * \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, &parameters, 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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