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.




git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2241 74dad513-b988-da41-8d7b-12977e46ad98

3 files changed, 36 insertions, 0 deletions

diff --git a/pjsip/include/pjsua-lib/pjsua_internal.h b/pjsip/include/pjsua-lib/pjsua_internal.h
index 5d7ff8f4..3839fa8f 100644
--- a/pjsip/include/pjsua-lib/pjsua_internal.h
+++ b/pjsip/include/pjsua-lib/pjsua_internal.h
@@ -66,6 +66,13 @@ typedef struct pjsua_call
     pjmedia_session	*session;   /**< The media session.		    */
     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
 					 call was triggered by xfer.	    */
diff --git a/pjsip/src/pjsua-lib/pjsua_call.c b/pjsip/src/pjsua-lib/pjsua_call.c
index 114da50d..737cc017 100644
--- a/pjsip/src/pjsua-lib/pjsua_call.c
+++ b/pjsip/src/pjsua-lib/pjsua_call.c
@@ -105,6 +105,9 @@ static void reset_call(pjsua_call_id id)
     call->session = NULL;
     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;
     call->conf_slot = PJSUA_INVALID_ID;
diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c
index 69314d30..6b0d2f63 100644
--- a/pjsip/src/pjsua-lib/pjsua_media.c
+++ b/pjsip/src/pjsua-lib/pjsua_media.c
@@ -1158,6 +1158,23 @@ static void stop_media_session(pjsua_call_id call_id)
     }
 
     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);
 	}
@@ -1322,6 +1339,15 @@ pj_status_t pjsua_media_channel_update(pjsua_call_id call_id,
 	/* 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. */
 	status = pjmedia_session_create( pjsua_var.med_endpt, &sess_info,
 					 &call->med_tp,