>From 97e92bd397b9a35f73b000368774968e89edf431 Mon Sep 17 00:00:00 2001
From: Pedro Sanchez <address@hidden>
Date: Fri, 11 Feb 2011 15:02:13 -0500
Subject: [PATCH] Make RTP traffic timer start when SIP negotiation starts.

---
 mediastreamer2/src/audiostream.c |   54 +++++++++++++++++++++----------------
 1 files changed, 31 insertions(+), 23 deletions(-)

diff --git a/mediastreamer2/src/audiostream.c b/mediastreamer2/src/audiostream.c
index 5ca1c9e..a2123b0 100644
--- a/mediastreamer2/src/audiostream.c
+++ b/mediastreamer2/src/audiostream.c
@@ -144,27 +144,30 @@ ms_time (time_t *t)
 }
 #endif
 
+static void _reset_rtp_alive_info(AudioStream * stream){
+	stream->last_packet_count=0;
+	stream->last_packet_time=time(NULL);
+}
+
 bool_t audio_stream_alive(AudioStream * stream, int timeout){
 	RtpSession *session=stream->session;
 	const rtp_stats_t *stats=rtp_session_get_stats(session);
-	if (stats->recv!=0){
-		if (stream->evq){
-			OrtpEvent *ev=ortp_ev_queue_get(stream->evq);
-			if (ev!=NULL){
-				if (ortp_event_get_type(ev)==ORTP_EVENT_RTCP_PACKET_RECEIVED){
-					stream->last_packet_time=ms_time(NULL);
-				}
-				ortp_event_destroy(ev);
+	if (stats->recv && stream->evq){
+		OrtpEvent *ev=ortp_ev_queue_get(stream->evq);
+		if (ev!=NULL){
+			if (ortp_event_get_type(ev)==ORTP_EVENT_RTCP_PACKET_RECEIVED){
+				stream->last_packet_time=ms_time(NULL);
 			}
+			ortp_event_destroy(ev);
 		}
-		if (stats->recv!=stream->last_packet_count){
-			stream->last_packet_count=stats->recv;
-			stream->last_packet_time=ms_time(NULL);
-		}else{
-			if (ms_time(NULL)-stream->last_packet_time>timeout){
-				/* more than timeout seconds of inactivity*/
-				return FALSE;
-			}
+	}
+	if (stats->recv!=stream->last_packet_count){
+		stream->last_packet_count=stats->recv;
+		stream->last_packet_time=ms_time(NULL);
+	}else{
+		if (ms_time(NULL)-stream->last_packet_time>timeout){
+			/* more than timeout seconds of inactivity*/
+			return FALSE;
 		}
 	}
 	return TRUE;
@@ -206,6 +209,7 @@ static void payload_type_changed(RtpSession *session, unsigned long data){
 	AudioStream *stream=(AudioStream*)data;
 	int pt=rtp_session_get_recv_payload_type(stream->session);
 	audio_stream_change_decoder(stream,pt);
+	_reset_rtp_alive_info(stream); 
 }
 
 int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char *remip,int remport,
@@ -379,6 +383,9 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	ms_ticker_attach(stream->ticker,stream->soundread);
 	ms_ticker_attach(stream->ticker,stream->rtprecv);
 
+	_reset_rtp_alive_info(stream);
+
+
 	return 0;
 }
 
@@ -463,6 +470,7 @@ AudioStream *audio_stream_new(int locport, bool_t ipv6){
 	stream->use_gc=FALSE;
 	stream->use_agc=FALSE;
 	stream->use_ng=FALSE;
+	_reset_rtp_alive_info(stream);
 	ms_filter_enable_statistics(TRUE);
 	ms_filter_reset_statistics();
 	return stream;
@@ -586,7 +594,7 @@ void audio_stream_stop(AudioStream * stream)
 }
 
 RingStream * ring_start(const char *file, int interval, MSSndCard *sndcard){
-   return ring_start_with_cb(file,interval,sndcard,NULL,NULL);
+	return ring_start_with_cb(file,interval,sndcard,NULL,NULL);
 }
 
 RingStream * ring_start_with_cb(const char *file,int interval,MSSndCard *sndcard, MSFilterNotifyFunc func,void * user_data)
@@ -687,10 +695,10 @@ void audio_stream_get_local_rtp_stats(AudioStream *stream, rtp_stats_t *lstats){
 
 void audio_stream_mute_rtp(AudioStream *stream, bool_t val) 
 {
-  if (stream->rtpsend){
-    if (val)
-      ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_MUTE_MIC,&val);
-    else
-      ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_UNMUTE_MIC,&val);
-  }
+	if (stream->rtpsend){
+		if (val)
+			ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_MUTE_MIC,&val);
+		else
+			ms_filter_call_method(stream->rtpsend,MS_RTP_SEND_UNMUTE_MIC,&val);
+	}
 }
-- 
1.7.0.4