Changeset 1928

Timestamp:

Apr 14, 2008 2:57:29 AM (16 years ago)

Author:

bennylp

Message:

Ticket #525: Crash on call update or re-invite (Thanks Alexey)

File:

• pjproject/trunk/pjsip/src/pjsua-lib/pjsua_media.c (modified) (2 diffs)

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

@@ -783 +783 @@
     }
 
-    /* Stop media transport (for good measure!) */
-    pjmedia_transport_media_stop(call->med_tp);
-
-    /* Close upper entry of transport stack */
-    if (call->med_orig && (call->med_tp != call->med_orig)) {
-        pjmedia_transport_close(call->med_tp);
-        call->med_tp = call->med_orig;
-    }
-
 #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-    /* Check if SRTP requires secure signaling */
-    if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) {
-        if (security_level < acc->cfg.srtp_secure_signaling) {
+    /* This function may be called when SRTP transport already exists 
+     * (e.g: in re-invite, update), don't need to destroy/re-create.
+     */
+    if (!call->med_orig || call->med_tp == call->med_orig) {
+
+        /* Check if SRTP requires secure signaling */
+        if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) {
+            if (security_level < acc->cfg.srtp_secure_signaling) {
+                if (sip_err_code)
+                    *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+                return PJSIP_ESESSIONINSECURE;
+            }
+        }
+
+        /* Always create SRTP adapter */
+        pjmedia_srtp_setting_default(&srtp_opt);
+        srtp_opt.close_member_tp = PJ_FALSE;
+        srtp_opt.use = acc->cfg.use_srtp;
+        status = pjmedia_transport_srtp_create(pjsua_var.med_endpt, 
+                                               call->med_tp,
+                                               &srtp_opt, &srtp);
+        if (status != PJ_SUCCESS) {
             if (sip_err_code)
-                *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
-            return PJSIP_ESESSIONINSECURE;
-        }
-    }
-
-    /* Always create SRTP adapter */
-    pjmedia_srtp_setting_default(&srtp_opt);
-    srtp_opt.close_member_tp = PJ_FALSE;
-    srtp_opt.use = acc->cfg.use_srtp;
-    status = pjmedia_transport_srtp_create(pjsua_var.med_endpt, 
-                                           call->med_tp,
-                                           &srtp_opt, &srtp);
-    if (status != PJ_SUCCESS) {
-        if (sip_err_code)
-            *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
-        return status;
-    }
-
-    /* Set SRTP as current media transport */
-    call->med_orig = call->med_tp;
-    call->med_tp = srtp;
+                *sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
+            return status;
+        }
+
+        /* Set SRTP as current media transport */
+        call->med_orig = call->med_tp;
+        call->med_tp = srtp;
+    }
 #else
     call->med_orig = call->med_tp;
@@ -902 +899 @@
 {
     pjsua_call *call = &pjsua_var.calls[call_id];
+
+    pjmedia_transport_media_stop(call->med_tp);
 
     if (call->conf_slot != PJSUA_INVALID_ID) {