Changeset 4338

Timestamp:

Jan 30, 2013 4:04:23 AM (12 years ago)

Author:

nanang

Message:

Re #1568: (only for 1.x) media transport should be (re)started even when media is unchanged.

File:

• pjproject/branches/1.x/pjsip/src/pjsua-lib/pjsua_media.c (modified) (4 diffs)

pjproject/branches/1.x/pjsip/src/pjsua-lib/pjsua_media.c

@@ -1710 +1710 @@
 
 
+static pj_status_t audio_channel_update(pjsua_call_id call_id,
+                                        int prev_media_st,
+                                        pjmedia_session_info *sess_info,
+                                        const pjmedia_sdp_session *local_sdp,
+                                        const pjmedia_sdp_session *remote_sdp)
+{
+    pjsua_call *call = &pjsua_var.calls[call_id];
+    pjmedia_stream_info *si = &sess_info->stream_info[0];
+    pjmedia_port *media_port;
+    pj_status_t status;
+    
+    /* Optionally, application may modify other stream settings here
+     * (such as jitter buffer parameters, codec ptime, etc.)
+     */
+    si->jb_init = pjsua_var.media_cfg.jb_init;
+    si->jb_min_pre = pjsua_var.media_cfg.jb_min_pre;
+    si->jb_max_pre = pjsua_var.media_cfg.jb_max_pre;
+    si->jb_max = pjsua_var.media_cfg.jb_max;
+    
+    /* Set SSRC */
+    si->ssrc = call->ssrc;
+    
+    /* Set RTP timestamp & sequence, normally these value are intialized
+     * automatically when stream session created, but for some cases (e.g:
+     * call reinvite, call update) timestamp and sequence need to be kept
+     * contigue.
+     */
+    si->rtp_ts = call->rtp_tx_ts;
+    si->rtp_seq = call->rtp_tx_seq;
+    si->rtp_seq_ts_set = call->rtp_tx_seq_ts_set;
+    
+    #if defined(PJMEDIA_STREAM_ENABLE_KA) && PJMEDIA_STREAM_ENABLE_KA!=0
+    /* Enable/disable stream keep-alive and NAT hole punch. */
+    si->use_ka = pjsua_var.acc[call->acc_id].cfg.use_stream_ka;
+    #endif
+    
+    /* Create session based on session info. */
+    status = pjmedia_session_create( pjsua_var.med_endpt, sess_info,
+                                     &call->med_tp,
+                                     call, &call->session );
+    if (status != PJ_SUCCESS) {
+        return status;
+    }
+    
+    if (prev_media_st == PJSUA_CALL_MEDIA_NONE)
+        pjmedia_session_send_rtcp_sdes(call->session);
+    
+    /* If DTMF callback is installed by application, install our
+     * callback to the session.
+     */
+    if (pjsua_var.ua_cfg.cb.on_dtmf_digit) {
+        pjmedia_session_set_dtmf_callback(call->session, 0, 
+                                          &dtmf_callback, 
+                                          (void*)(long)(call->index));
+    }
+    
+    /* Get the port interface of the first stream in the session.
+     * We need the port interface to add to the conference bridge.
+     */
+    pjmedia_session_get_port(call->session, 0, &media_port);
+    
+    /* Notify application about stream creation.
+     * Note: application may modify media_port to point to different
+     * media port
+     */
+    if (pjsua_var.ua_cfg.cb.on_stream_created) {
+        pjsua_var.ua_cfg.cb.on_stream_created(call_id, call->session,
+                                              0, &media_port);
+    }
+    
+    /*
+     * Add the call to conference bridge.
+     */
+    {
+        char tmp[PJSIP_MAX_URL_SIZE];
+        pj_str_t port_name;
+    
+        port_name.ptr = tmp;
+        port_name.slen = pjsip_uri_print(PJSIP_URI_IN_REQ_URI,
+                                         call->inv->dlg->remote.info->uri,
+                                         tmp, sizeof(tmp));
+        if (port_name.slen < 1) {
+            port_name = pj_str("call");
+        }
+        status = pjmedia_conf_add_port( pjsua_var.mconf, 
+                                        call->inv->pool,
+                                        media_port, 
+                                        &port_name,
+                                        (unsigned*)&call->conf_slot);
+        if (status != PJ_SUCCESS) {
+            return status;
+        }
+    }
+    
+    return PJ_SUCCESS;
+}
+
+
 /* Internal function: update media channel after SDP negotiation.
  * Warning: do not use temporary/flip-flop pool, e.g: inv->pool_prov,
@@ -1721 +1819 @@
 {
     unsigned i;
-    int prev_media_st = 0;
+    int prev_media_st;
     pjsua_call *call = &pjsua_var.calls[call_id];
     pjmedia_session_info sess_info;
     pjmedia_stream_info *si = NULL;
-    pjmedia_port *media_port;
     pj_status_t status;
+    pj_bool_t media_changed = PJ_FALSE;
     int audio_idx;
 
@@ -1758 +1856 @@
     audio_idx = find_audio_index(local_sdp, PJ_TRUE);
 
+    /* Get previous media status */
+    prev_media_st = call->media_st;
+    
     /* Check if media has just been changed. */
-    if (!pjsua_var.media_cfg.no_smart_media_update &&
-        !is_media_changed(call, audio_idx, &sess_info))
+    if (pjsua_var.media_cfg.no_smart_media_update ||
+        is_media_changed(call, audio_idx, &sess_info))
     {
+        media_changed = PJ_TRUE;
+
+        /* Destroy existing media session, if any. */
+        stop_media_session(call->index);
+    } else {
         PJ_LOG(4,(THIS_FILE, "Media session for call %d is unchanged",
                              call_id));
-        return PJ_SUCCESS;
-    }
-
-    /* Destroy existing media session, if any. */
-    prev_media_st = call->media_st;
-    stop_media_session(call->index);
+    }
+
 
     for (i = 0; i < sess_info.stream_cnt; ++i) {
@@ -1847 +1949 @@
         }
 
-
-        /* Optionally, application may modify other stream settings here
-         * (such as jitter buffer parameters, codec ptime, etc.)
-         */
-        si->jb_init = pjsua_var.media_cfg.jb_init;
-        si->jb_min_pre = pjsua_var.media_cfg.jb_min_pre;
-        si->jb_max_pre = pjsua_var.media_cfg.jb_max_pre;
-        si->jb_max = pjsua_var.media_cfg.jb_max;
-
-        /* Set SSRC */
-        si->ssrc = call->ssrc;
-
-        /* Set RTP timestamp & sequence, normally these value are intialized
-         * automatically when stream session created, but for some cases (e.g:
-         * call reinvite, call update) timestamp and sequence need to be kept
-         * contigue.
-         */
-        si->rtp_ts = call->rtp_tx_ts;
-        si->rtp_seq = call->rtp_tx_seq;
-        si->rtp_seq_ts_set = call->rtp_tx_seq_ts_set;
-
-#if defined(PJMEDIA_STREAM_ENABLE_KA) && PJMEDIA_STREAM_ENABLE_KA!=0
-        /* Enable/disable stream keep-alive and NAT hole punch. */
-        si->use_ka = pjsua_var.acc[call->acc_id].cfg.use_stream_ka;
-#endif
-
-        /* Create session based on session info. */
-        status = pjmedia_session_create( pjsua_var.med_endpt, &sess_info,
-                                         &call->med_tp,
-                                         call, &call->session );
-        if (status != PJ_SUCCESS) {
-            return status;
-        }
-
-        if (prev_media_st == PJSUA_CALL_MEDIA_NONE)
-            pjmedia_session_send_rtcp_sdes(call->session);
-
-        /* If DTMF callback is installed by application, install our
-         * callback to the session.
-         */
-        if (pjsua_var.ua_cfg.cb.on_dtmf_digit) {
-            pjmedia_session_set_dtmf_callback(call->session, 0, 
-                                              &dtmf_callback, 
-                                              (void*)(long)(call->index));
-        }
-
-        /* Get the port interface of the first stream in the session.
-         * We need the port interface to add to the conference bridge.
-         */
-        pjmedia_session_get_port(call->session, 0, &media_port);
-
-        /* Notify application about stream creation.
-         * Note: application may modify media_port to point to different
-         * media port
-         */
-        if (pjsua_var.ua_cfg.cb.on_stream_created) {
-            pjsua_var.ua_cfg.cb.on_stream_created(call_id, call->session,
-                                                  0, &media_port);
-        }
-
-        /*
-         * Add the call to conference bridge.
-         */
-        {
-            char tmp[PJSIP_MAX_URL_SIZE];
-            pj_str_t port_name;
-
-            port_name.ptr = tmp;
-            port_name.slen = pjsip_uri_print(PJSIP_URI_IN_REQ_URI,
-                                             call->inv->dlg->remote.info->uri,
-                                             tmp, sizeof(tmp));
-            if (port_name.slen < 1) {
-                port_name = pj_str("call");
-            }
-            status = pjmedia_conf_add_port( pjsua_var.mconf, 
-                                            call->inv->pool,
-                                            media_port, 
-                                            &port_name,
-                                            (unsigned*)&call->conf_slot);
-            if (status != PJ_SUCCESS) {
+        if (media_changed) {
+            status = audio_channel_update(call_id, prev_media_st, &sess_info,
+                                          local_sdp, remote_sdp);
+            if (status != PJ_SUCCESS)
                 return status;
-            }
         }