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

Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
Session-Timers for SIP over TCP (and for SIP over TLS).

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 dropped calls. Consequently even with this change,
you might be better-off going for session-timers=refuse in your sip.conf.

ASTERISK-19968 #close

Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957

1 files changed, 20 insertions, 32 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index bf387bccd..4be53c218 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4200,19 +4200,6 @@ 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;
@@ -4249,6 +4236,10 @@ 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);
@@ -24592,6 +24583,7 @@ 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 = "";
@@ -24620,28 +24612,24 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
 			ast_channel_hangupcause_set(owner, hangup_sip2cause(resp));
 	}
 
-	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 */