Changeset 3841 for pjproject/trunk/pjsip/src/pjsua-lib/pjsua_media.c

Timestamp:

Oct 24, 2011 9:28:13 AM (13 years ago)

Author:

ming

Message:

Re #1395: Backport of PJSIP 1.x branch into PJSIP 2.0 trunk

• Backport of r3557:r3832

TODO: ticket #1268 (Option for automatic/manual sending of RTCP SDES/BYE for the stream) for video stream.

Location:

pjproject/trunk

Files:

• . (modified) (1 prop)
• pjsip/src/pjsua-lib/pjsua_media.c (modified) (10 diffs)

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

@@ -397 +397 @@
  * Destroy pjsua media subsystem.
  */
-pj_status_t pjsua_media_subsys_destroy(void)
+pj_status_t pjsua_media_subsys_destroy(unsigned flags)
 {
     unsigned i;
@@ -442 +442 @@
             }
             if (call_med->tp && call_med->tp_auto_del) {
+                /* TODO: check if we're not allowed to send to network in the
+                 *       "flags", and if so do not do TURN allocation...
+                 */
+                PJ_UNUSED_ARG(flags);
                 pjmedia_transport_close(call_med->tp);
             }
@@ -1295 +1299 @@
         set_media_tp_state(call_med, PJSUA_MED_TP_IDLE);
 
+    if (!call_med->tp_orig &&
+        pjsua_var.ua_cfg.cb.on_create_media_transport)
+    {
+        call_med->use_custom_med_tp = PJ_TRUE;
+    } else
+        call_med->use_custom_med_tp = PJ_FALSE;
+
 #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
     /* 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->tp_orig || call_med->tp == call_med->tp_orig) {
+    if (!call_med->tp_orig) {
         pjmedia_srtp_setting srtp_opt;
         pjmedia_transport *srtp = NULL;
@@ -1315 +1326 @@
         pjmedia_srtp_setting_default(&srtp_opt);
         srtp_opt.close_member_tp = PJ_TRUE;
-        /* If media session has been ever established, let's use remote's
+        /* If media session has been ever established, let's use remote's 
          * preference in SRTP usage policy, especially when it is stricter.
          */
@@ -1520 +1531 @@
             }
 
-            status = pjmedia_transport_media_create(
-                         call_med->tp, tmp_pool,
-                         0, call->async_call.rem_sdp, mi);
+            if (call_med->use_custom_med_tp) {
+                unsigned custom_med_tp_flags = 0;
+
+                /* Use custom media transport returned by the application */
+                call_med->tp =
+                    (*pjsua_var.ua_cfg.cb.on_create_media_transport)
+                        (call_id, mi, call_med->tp,
+                         custom_med_tp_flags);
+                if (!call_med->tp) {
+                    status =
+                        PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE);
+                }
+            }
+
+            if (call_med->tp) {
+                status = pjmedia_transport_media_create(
+                             call_med->tp, tmp_pool,
+                             0, call->async_call.rem_sdp, mi);
+            }
             if (status != PJ_SUCCESS) {
                 call->med_ch_info.status = status;
@@ -2105 +2132 @@
         PJ_LOG(4,(THIS_FILE, "Media session call%02d:%d is destroyed",
                              call_id, mi));
+        call_med->prev_state = call_med->state;
         call_med->state = PJSUA_CALL_MEDIA_NONE;
     }
@@ -2134 +2162 @@
     pj_log_push_indent();
 
-    stop_media_session(call_id);
-
     for (mi=0; mi<call->med_cnt; ++mi) {
         pjsua_call_media *call_med = &call->media[mi];
 
-        if (call_med->tp_st > PJSUA_MED_TP_IDLE) {
+        if (call_med->type == PJMEDIA_TYPE_AUDIO && call_med->strm.a.stream)
+            pjmedia_stream_send_rtcp_bye(call_med->strm.a.stream);
+    }
+
+    stop_media_session(call_id);
+
+    for (mi=0; mi<call->med_cnt; ++mi) {
+        pjsua_call_media *call_med = &call->media[mi];
+
+        if (call_med->tp_st > PJSUA_MED_TP_IDLE) {
             pjmedia_transport_media_stop(call_med->tp);
             set_media_tp_state(call_med, PJSUA_MED_TP_IDLE);
@@ -2154 +2189 @@
             call_med->tp = call_med->tp_orig = NULL;
         }
+        call_med->tp_orig = NULL;
     }
 
@@ -2206 +2242 @@
     if (status != PJ_SUCCESS)
         goto on_return;
+
+    si->rtcp_sdes_bye_disabled = PJ_TRUE;
 
     /* Check if no media is active */
@@ -2296 +2334 @@
             goto on_return;
         }
+
+        if (call_med->prev_state == PJSUA_CALL_MEDIA_NONE)
+            pjmedia_stream_send_rtcp_sdes(call_med->strm.a.stream);
 
         /* If DTMF callback is installed by application, install our