Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."

2 files changed, 28 insertions, 32 deletions

diff --git a/CHANGES b/CHANGES
index d5d256ef9..31b521c90 100644
--- a/CHANGES
+++ b/CHANGES
@@ -181,6 +181,14 @@ chan_sip
    NOTE: This is again separated by an exclamation mark, so the To: header may
    not contain one of those.
 
+ * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
+   Previously Asterisk dropped calls only with UDP transports. However with
+   longer international calls via TCP, the SIP channel might break, because
+   all hops on the Internet route must stay online (have not a single power
+   outage, for example). Therefore with Session-Timers enabled (which are
+   enabled at default), you might see additional dropped calls. Consequently
+   please, consider to go for session-timers=refuse in your sip.conf.
+
 chan_pjsip
 ------------------
  * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6913a039b..5f422d72a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4205,19 +4205,6 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, in
 		p->pendinginvite = seqno;
 	}
 
-	/* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
-	/* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
-	/*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
-	if (!(p->socket.type & AST_TRANSPORT_UDP)) {
-		xmitres = __sip_xmit(p, data);	/* Send packet */
-		if (xmitres == XMIT_ERROR) {	/* Serious network trouble, no need to try again */
-			append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
-			return AST_FAILURE;
-		} else {
-			return AST_SUCCESS;
-		}
-	}
-
 	pkt = ao2_alloc_options(sizeof(*pkt), sip_pkt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
 	if (!pkt) {
 		return AST_FAILURE;
@@ -4254,6 +4241,10 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, in
 	pkt->time_sent = ast_tvnow(); /* time packet was sent */
 	pkt->retrans_stop_time = 64 * (pkt->timer_t1 ? pkt->timer_t1 : DEFAULT_TIMER_T1); /* time in ms after pkt->time_sent to stop retransmission */
 
+	if (!(p->socket.type & AST_TRANSPORT_UDP)) {
+		pkt->retrans_stop = 1;
+	}
+
 	/* Schedule retransmission */
 	ao2_t_ref(pkt, +1, "Schedule packet retransmission");
 	pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
@@ -24654,6 +24645,7 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
 	char *c_copy = ast_strdupa(c);
 	/* Skip the Cseq and its subsequent spaces */
 	const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
+	int ack_res = FALSE;
 
 	if (!msg)
 		msg = "";
@@ -24668,28 +24660,24 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
 		}
 	}
 
-	if (p->socket.type == AST_TRANSPORT_UDP) {
-		int ack_res = FALSE;
-
-		/* Acknowledge whatever it is destined for */
-		if ((resp >= 100) && (resp <= 199)) {
-			/* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
-			if (sipmethod == SIP_INVITE) {
-				ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
-			}
-		} else {
-			ack_res = __sip_ack(p, seqno, 0, sipmethod);
+	/* Acknowledge whatever it is destined for */
+	if ((resp >= 100) && (resp <= 199)) {
+		/* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
+		if (sipmethod == SIP_INVITE) {
+			ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
 		}
+	} else {
+		ack_res = __sip_ack(p, seqno, 0, sipmethod);
+	}
 
-		if (ack_res == FALSE) {
-			/* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
-			if (sipmethod == SIP_INVITE && resp >= 200) {
-				transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
-			}
-
-			append_history(p, "Ignore", "Ignoring this retransmit\n");
-			return;
+	if (ack_res == FALSE) {
+		/* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
+		if (sipmethod == SIP_INVITE && resp >= 200) {
+			transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
 		}
+
+		append_history(p, "Ignore", "Ignoring this retransmit\n");
+		return;
 	}
 
 	/* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */