Changeset 5928 for pjproject/trunk/pjsip/src/pjsua-lib/pjsua_call.c

Timestamp:

Jan 8, 2019 9:43:21 AM (6 years ago)

Author:

riza

Message:

Close #2173: On PJSUA2, application will not be notified when when SDP nego fails due to unsupported codec.

File:

• pjproject/trunk/pjsip/src/pjsua-lib/pjsua_call.c (modified) (8 diffs)

pjproject/trunk/pjsip/src/pjsua-lib/pjsua_call.c

@@ -1085 +1085 @@
 }
 
+pj_status_t create_temp_sdp(pj_pool_t *pool,
+                            const pjmedia_sdp_session *rem_sdp,
+                            pjmedia_sdp_session **p_sdp)
+{
+    const pj_str_t STR_AUDIO = { "audio", 5 };
+    const pj_str_t STR_VIDEO = { "video", 5 };
+    const pj_str_t STR_IP6 = { "IP6", 3};
+
+    pjmedia_sdp_session *sdp;
+    pj_sockaddr origin;
+    pj_uint16_t tmp_port = 50123;
+    pj_status_t status = PJ_SUCCESS;
+    pj_str_t tmp_st;
+    unsigned i = 0;
+    pj_bool_t sess_use_ipv4 = PJ_TRUE;
+
+    /* Get one address to use in the origin field */
+    pj_sockaddr_init(PJ_AF_INET, &origin, pj_strset2(&tmp_st, "127.0.0.1"), 0);
+
+    /* Create the base (blank) SDP */
+    status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, pool, NULL,
+                                           &origin, &sdp);
+    if (status != PJ_SUCCESS)
+        return status;
+
+    if (rem_sdp->conn && pj_stricmp(&rem_sdp->conn->addr_type, &STR_IP6)==0) {
+        sess_use_ipv4 = PJ_FALSE;
+    }
+
+    for (; i< rem_sdp->media_count ; ++i) {
+        pjmedia_sdp_media *m = NULL;
+        pjmedia_sock_info sock_info;
+        pj_bool_t med_use_ipv4 = sess_use_ipv4;
+
+        if (rem_sdp->media[i]->conn && 
+            pj_stricmp(&rem_sdp->media[i]->conn->addr_type, &STR_IP6) == 0) 
+        {
+            med_use_ipv4 = PJ_FALSE;
+        }
+
+        pj_sockaddr_init(med_use_ipv4?PJ_AF_INET:PJ_AF_INET6, 
+                         &sock_info.rtp_addr_name, 
+                         med_use_ipv4?pj_strset2(&tmp_st, "127.0.0.1"):
+                                      pj_strset2(&tmp_st, "::1"), 
+                         tmp_port++);
+
+        pj_sockaddr_init(med_use_ipv4?PJ_AF_INET:PJ_AF_INET6, 
+                         &sock_info.rtcp_addr_name, 
+                         med_use_ipv4?pj_strset2(&tmp_st, "127.0.0.1"):
+                                      pj_strset2(&tmp_st, "::1"), 
+                         tmp_port++);
+
+        if (pj_stricmp(&rem_sdp->media[i]->desc.media, &STR_AUDIO)==0) {
+            m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
+            status = pjmedia_endpt_create_audio_sdp(pjsua_var.med_endpt,
+                                                    pool, &sock_info, 0, &m);
+
+            if (status != PJ_SUCCESS)
+                return status;
+        
+        } else if (pj_stricmp(&rem_sdp->media[i]->desc.media, &STR_VIDEO)==0) {
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+            m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
+
+            pj_sockaddr_set_port(&sock_info.rtp_addr_name, ++tmp_port);
+            status = pjmedia_endpt_create_video_sdp(pjsua_var.med_endpt, pool,
+                                                    &sock_info, 0, &m);
+            if (status != PJ_SUCCESS)
+                return status;
+#else       
+            m = pjmedia_sdp_media_clone_deactivate(pool, rem_sdp->media[i]);
+#endif      
+        } else {
+            m = pjmedia_sdp_media_clone_deactivate(pool, rem_sdp->media[i]);
+        }
+        if (status != PJ_SUCCESS)
+            return status;
+        sdp->media[sdp->media_count++] = m;
+    }      
+
+    *p_sdp = sdp;
+    return PJ_SUCCESS;
+}
+
+static pj_status_t verify_request(const pjsua_call *call,
+                                  pjsip_rx_data *rdata,
+                                  pj_bool_t use_tmp_sdp,
+                                  int *sip_err_code,
+                                  pjsip_tx_data **response)
+{
+    const pjmedia_sdp_session *offer = NULL;
+    pjmedia_sdp_session *answer;    
+    int err_code = 0;
+    pj_status_t status;
+    
+    /* Get remote SDP offer (if any). */
+    if (call->inv->neg)
+    {
+        pjmedia_sdp_neg_get_neg_remote(call->inv->neg, &offer);
+    }
+
+    if (use_tmp_sdp) {
+        if (offer == NULL)
+            return PJ_SUCCESS;
+
+        /* Create temporary SDP to check for codec support and capability 
+         * to handle the required SIP extensions.
+         */
+        status = create_temp_sdp(call->inv->pool_prov, offer, &answer);
+
+        if (status != PJ_SUCCESS) {
+            err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
+            pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
+        }
+    } else {
+        status = pjsua_media_channel_create_sdp(call->index,
+                                                call->async_call.dlg->pool,
+                                                offer, &answer, sip_err_code);
+
+        if (status != PJ_SUCCESS) {
+            pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
+        } else {
+            status = pjsip_inv_set_local_sdp(call->inv, answer);
+            if (status != PJ_SUCCESS) {
+                pjsua_perror(THIS_FILE, "Error setting local SDP", status);
+                err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;                
+            }
+        }
+    }
+
+    if (status == PJ_SUCCESS) {
+        unsigned options = 0;
+
+        /* Verify that we can handle the request. */
+        status = pjsip_inv_verify_request3(rdata,
+                                           call->inv->pool_prov, &options, 
+                                           offer, answer, NULL, 
+                                           pjsua_var.endpt, response);
+        if (status != PJ_SUCCESS) {
+            /*
+             * No we can't handle the incoming INVITE request.
+             */
+            if (response)
+                err_code = (*response)->msg->line.status.code;
+            else
+                err_code = PJSIP_ERRNO_TO_SIP_STATUS(status);
+        }
+    }
+    if (sip_err_code)
+        *sip_err_code = err_code;
+
+    return status;
+}
 
 /* Incoming call callback when media transport creation is completed. */
 static pj_status_t
-on_incoming_call_med_tp_complete(pjsua_call_id call_id,
-                                 const pjsua_med_tp_state_info *info)
+on_incoming_call_med_tp_complete2(pjsua_call_id call_id,
+                                  const pjsua_med_tp_state_info *info,
+                                  pjsip_rx_data *rdata,
+                                  int *sip_err_code,
+                                  pjsip_tx_data **tdata)
 {
     pjsua_call *call = &pjsua_var.calls[call_id];
-    const pjmedia_sdp_session *offer=NULL;
-    pjmedia_sdp_session *answer;
+    pjsip_dialog *dlg = call->async_call.dlg;    
+    pj_status_t status = (info? info->status: PJ_SUCCESS);
+    int err_code = (info? info->sip_err_code: 0);
     pjsip_tx_data *response = NULL;
-    unsigned options = 0;
-    pjsip_dialog *dlg = call->async_call.dlg;
-    int sip_err_code = (info? info->sip_err_code: 0);
-    pj_status_t status = (info? info->status: PJ_SUCCESS);
 
     PJSUA_LOCK();
-    
+
+    if (sip_err_code)
+        *sip_err_code = err_code;
+
     /* Increment the dialog's lock to prevent it to be destroyed prematurely,
      * such as in case of transport error.
@@ -1124 +1280 @@
     }
 
-    /* Get remote SDP offer (if any). */
-    if (call->inv->neg)
-        pjmedia_sdp_neg_get_neg_remote(call->inv->neg, &offer);
-
-    status = pjsua_media_channel_create_sdp(call_id,
-                                            call->async_call.dlg->pool,
-                                            offer, &answer, &sip_err_code);
-    if (status != PJ_SUCCESS) {
-        pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
-        goto on_return;
-    }
-
-    status = pjsip_inv_set_local_sdp(call->inv, answer);
-    if (status != PJ_SUCCESS) {
-        pjsua_perror(THIS_FILE, "Error setting local SDP", status);
-        sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
-        goto on_return;
-    }
-
-    /* Verify that we can handle the request. */
-    status = pjsip_inv_verify_request3(NULL,
-                                       call->inv->pool_prov, &options, offer,
-                                       answer, NULL, pjsua_var.endpt, &response);
-    if (status != PJ_SUCCESS) {
-        /*
-         * No we can't handle the incoming INVITE request.
-         */
-        sip_err_code = PJSIP_ERRNO_TO_SIP_STATUS(status);
-        goto on_return;
-    }
+    status = verify_request(call, rdata, PJ_FALSE, &err_code, &response);
 
 on_return:
@@ -1161 +1288 @@
          * will be terminated later, otherwise we end the session here.
          */
-        if (call->inv->state > PJSIP_INV_STATE_NULL) {
-            pjsip_tx_data *tdata;
-            pj_status_t status_;
-
-            status_ = pjsip_inv_end_session(call->inv, sip_err_code, NULL,
-                                            &tdata);
-            if (status_ == PJ_SUCCESS && tdata)
-                status_ = pjsip_inv_send_msg(call->inv, tdata);
-        }
-
+        if (call->inv->state > PJSIP_INV_STATE_NULL) {            
+            pj_status_t status_ = PJ_SUCCESS;
+
+            if (response == NULL) {
+                status_ = pjsip_inv_end_session(call->inv, err_code, NULL,
+                                                &response);
+            }
+
+            if (status_ == PJ_SUCCESS && response)
+                status_ = pjsip_inv_send_msg(call->inv, response);
+        }
         pjsua_media_channel_deinit(call->index);
     }
@@ -1193 +1321 @@
     
     pjsip_dlg_dec_lock(dlg);
+
+    if (sip_err_code)
+        *sip_err_code = err_code;
+
+    if (tdata)
+        *tdata = response;
     
     PJSUA_UNLOCK();
     return status;
+}
+
+static pj_status_t
+on_incoming_call_med_tp_complete(pjsua_call_id call_id,
+                                 const pjsua_med_tp_state_info *info)
+{
+    return on_incoming_call_med_tp_complete2(call_id, info, NULL, NULL, NULL);
 }
 
@@ -1611 +1752 @@
      */
     if (offer || replaced_dlg) {
+
+        /* This is only for initial verification, it will check the SDP for
+         * codec support and the capability to handle the required
+         * SIP extensions.
+         */
+        status = verify_request(call, rdata, PJ_TRUE, &sip_err_code, 
+                                &response);
+
+        if (status != PJ_SUCCESS) {
+            pjsip_dlg_inc_lock(dlg);
+
+            if (response) {
+                pjsip_dlg_send_response(dlg, call->inv->invite_tsx, response);
+
+            } else {
+                pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL);
+            }
+                
+            if (call->inv && call->inv->dlg) {
+                pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE);
+            }
+            pjsip_dlg_dec_lock(dlg);
+
+            call->inv = NULL;
+            call->async_call.dlg = NULL;
+            goto on_return;
+        }
         status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAS,
                                           call->secure_level,
@@ -1618 +1786 @@
                                           &on_incoming_call_med_tp_complete);
         if (status == PJ_SUCCESS) {
-            status = on_incoming_call_med_tp_complete(call_id, NULL);
-            if (status != PJ_SUCCESS) {
-                sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+            status = on_incoming_call_med_tp_complete2(call_id, NULL, 
+                                                       rdata, &sip_err_code, 
+                                                       &response);
+            if (status != PJ_SUCCESS) {         
                 /* Since the call invite's state is still PJSIP_INV_STATE_NULL,
                  * the invite session was not ended in
@@ -1627 +1796 @@
                  */
                 pjsip_dlg_inc_lock(dlg);
-                pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL);
+
+                if (response) {
+                    pjsip_dlg_send_response(dlg, call->inv->invite_tsx, 
+                                            response);
+
+                } else {
+                    pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, 
+                                      NULL);
+                }               
+
                 if (call->inv && call->inv->dlg) {
                     pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE);
@@ -1734 +1912 @@
      * above)
      */
-    dlg->mod_data[pjsua_var.mod.id] = call;
-    inv->mod_data[pjsua_var.mod.id] = call;
-
-    ++pjsua_var.call_cnt;
+    if (dlg->mod_data[pjsua_var.mod.id] == NULL) {
+        /* In PJSUA2, on_incoming_call() may be called from 
+         * on_media_transport_created() hence this might already set
+         * to allow notification about fail events via on_call_state() and
+         * on_call_tsx_state().
+         */
+        dlg->mod_data[pjsua_var.mod.id] = call;
+        inv->mod_data[pjsua_var.mod.id] = call;
+        ++pjsua_var.call_cnt;
+    }
 
     /* Check if this request should replace existing call */