Changeset 2241

Timestamp:

Aug 26, 2008 4:51:28 PM (16 years ago)

Author:

nanang

Message:

Ticket #602:

• Introduced new API pjmedia_rtp_session_init2() to enable intializing RTP session with non-default initial settings
• Updated stream so it can be created with non-default initial RTP settings.
• Updated pjsua-lib to make sure RTP timestamp and sequence contigue when stream session is restarted.

Location:

pjproject/trunk

Files:

• pjmedia/include/pjmedia/rtcp.h (modified) (1 diff)
• pjmedia/include/pjmedia/rtp.h (modified) (2 diffs)
• pjmedia/include/pjmedia/stream.h (modified) (1 diff)
• pjmedia/src/pjmedia/rtp.c (modified) (1 diff)
• pjmedia/src/pjmedia/stream.c (modified) (2 diffs)
• pjsip/include/pjsua-lib/pjsua_internal.h (modified) (1 diff)
• pjsip/src/pjsua-lib/pjsua_call.c (modified) (1 diff)
• pjsip/src/pjsua-lib/pjsua_media.c (modified) (2 diffs)

pjproject/trunk/pjmedia/include/pjmedia/rtcp.h

@@ -211 +211 @@
     
     pj_math_stat             rtt;   /**< Round trip delay statistic(in usec)*/
+
+    pj_uint32_t              rtp_tx_last_ts; /**< Last TX RTP timestamp.    */
+    pj_uint16_t              rtp_tx_last_seq;/**< Last TX RTP sequence.     */
 };
 

pjproject/trunk/pjmedia/include/pjmedia/rtp.h

@@ -221 +221 @@
 };
 
+
+/**
+ * RTP session settings.
+ */
+typedef struct pjmedia_rtp_session_setting
+{
+    pj_uint8_t       flags;         /**< Bitmask flags to specify whether such
+                                         field is set. Bitmask contents are:
+                                         (bit #0 is LSB)
+                                         bit #0: default payload type
+                                         bit #1: sender SSRC
+                                         bit #2: sequence
+                                         bit #3: timestamp                  */
+    int              default_pt;    /**< Default payload type.              */
+    pj_uint32_t      sender_ssrc;   /**< Sender SSRC.                       */
+    pj_uint16_t      seq;           /**< Sequence.                          */
+    pj_uint32_t      ts;            /**< Timestamp.                         */
+} pjmedia_rtp_session_setting;
+
+
 /**
  * @see pjmedia_rtp_status
@@ -239 +259 @@
                                                int default_pt, 
                                                pj_uint32_t sender_ssrc );
+
+/**
+ * This function will initialize the RTP session according to given parameters
+ * defined in RTP session settings.
+ *
+ * @param ses           The session.
+ * @param settings      RTP session settings.
+ *
+ * @return              PJ_SUCCESS if successfull.
+ */
+PJ_DECL(pj_status_t) pjmedia_rtp_session_init2( 
+                                    pjmedia_rtp_session *ses,
+                                    pjmedia_rtp_session_setting settings);
+
 
 /**

pjproject/trunk/pjmedia/include/pjmedia/stream.h

@@ -113 +113 @@
     int                 rx_event_pt;/**< Incoming pt for telephone-events.  */
     pj_uint32_t         ssrc;       /**< RTP SSRC.                          */
+    pj_uint32_t         rtp_ts;     /**< Initial RTP timestamp.             */
+    pj_uint16_t         rtp_seq;    /**< Initial RTP sequence number.       */
+    pj_uint8_t          rtp_seq_ts_set;
+                                    /**< Bitmask flags if initial RTP sequence 
+                                         and/or timestamp for sender are set.
+                                         bit 0/LSB : sequence flag 
+                                         bit 1     : timestamp flag         */
     int                 jb_init;    /**< Jitter buffer init delay in msec.  
                                          (-1 for default).                  */

pjproject/trunk/pjmedia/src/pjmedia/rtp.c

@@ -85 +85 @@
 }
 
+PJ_DEF(pj_status_t) pjmedia_rtp_session_init2( 
+                                    pjmedia_rtp_session *ses,
+                                    pjmedia_rtp_session_setting settings)
+{
+    pj_status_t status;
+    int          pt = 0;
+    pj_uint32_t  sender_ssrc = 0;
+
+    if (settings.flags & 1)
+        pt = settings.default_pt;
+    if (settings.flags & 2)
+        sender_ssrc = settings.sender_ssrc;
+
+    status = pjmedia_rtp_session_init(ses, pt, sender_ssrc);
+    if (status != PJ_SUCCESS)
+        return status;
+
+    if (settings.flags & 4) {
+        ses->out_extseq = settings.seq;
+        ses->out_hdr.seq = pj_htons((pj_uint16_t)ses->out_extseq);
+    }
+    if (settings.flags & 8)
+        ses->out_hdr.ts = pj_htonl(settings.ts);
+
+    return PJ_SUCCESS;
+}
+
 
 PJ_DEF(pj_status_t) pjmedia_rtp_encode_rtp( pjmedia_rtp_session *ses, 

pjproject/trunk/pjmedia/src/pjmedia/stream.c

@@ -873 +873 @@
     /* Update stat */
     pjmedia_rtcp_tx_rtp(&stream->rtcp, frame_out.size);
+    stream->rtcp.stat.rtp_tx_last_ts = pj_ntohl(stream->enc->rtp.out_hdr.ts);
+    stream->rtcp.stat.rtp_tx_last_seq = pj_ntohs(stream->enc->rtp.out_hdr.seq);
 
     return PJ_SUCCESS;
@@ -1413 +1415 @@
     /* Create RTP and RTCP sessions: */
 
-    status = pjmedia_rtp_session_init(&channel->rtp, pt, param->ssrc);
+    if (param->rtp_seq_ts_set == 0) {
+        status = pjmedia_rtp_session_init(&channel->rtp, pt, param->ssrc);
+    } else {
+        pjmedia_rtp_session_setting settings;
+
+        settings.flags = (param->rtp_seq_ts_set << 2) | 3;
+        settings.default_pt = pt;
+        settings.sender_ssrc = param->ssrc;
+        settings.seq = param->rtp_seq;
+        settings.ts = param->rtp_ts;
+        status = pjmedia_rtp_session_init2(&channel->rtp, settings);
+    }
     if (status != PJ_SUCCESS)
         return status;

pjproject/trunk/pjsip/include/pjsua-lib/pjsua_internal.h

@@ -67 +67 @@
     int                  audio_idx; /**< Index of m=audio in SDP.           */
     pj_uint32_t          ssrc;      /**< RTP SSRC                           */
+    pj_uint32_t          rtp_tx_ts; /**< Initial RTP timestamp for sender.  */
+    pj_uint16_t          rtp_tx_seq;/**< Initial RTP sequence for sender.   */
+    pj_uint8_t           rtp_tx_seq_ts_set;
+                                    /**< Bitmask flags if initial RTP sequence
+                                         and/or timestamp for sender are set.
+                                         bit 0/LSB : sequence flag 
+                                         bit 1     : timestamp flag         */
     int                  conf_slot; /**< Slot # in conference bridge.       */
     pjsip_evsub         *xfer_sub;  /**< Xfer server subscription, if this

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

@@ -106 +106 @@
     call->audio_idx = -1;
     call->ssrc = pj_rand();
+    call->rtp_tx_seq = 0;
+    call->rtp_tx_ts = 0;
+    call->rtp_tx_seq_ts_set = 0;
     call->xfer_sub = NULL;
     call->last_code = (pjsip_status_code) 0;

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

@@ -1159 +1159 @@
 
     if (call->session) {
+        pjmedia_rtcp_stat stat;
+
+        if (pjmedia_session_get_stream_stat(call->session, 
+                                            call->audio_idx, 
+                                            &stat) == PJ_SUCCESS)
+        {
+            /* Save RTP timestamp & sequence, so when media session is 
+             * restarted, those values will be restored as the initial 
+             * RTP timestamp & sequence of the new media session. So in 
+             * the same call session, RTP timestamp and sequence are 
+             * guaranteed to be contigue.
+             */
+            call->rtp_tx_seq_ts_set = 1 | (1 << 1);
+            call->rtp_tx_seq = stat.rtp_tx_last_seq;
+            call->rtp_tx_ts = stat.rtp_tx_last_ts;
+        }
+
         if (pjsua_var.ua_cfg.cb.on_stream_destroyed) {
             pjsua_var.ua_cfg.cb.on_stream_destroyed(call_id, call->session, 0);
@@ -1322 +1339 @@
         /* 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;
 
         /* Create session based on session info. */