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

CHANGES

@@ -18,6 +18,16 @@ chan_dahdi
    The option determines how many seconds into a call before faxdetect
    is disabled for the call.  Setting the value to zero disables the timeout.
 
+chan_sip
+------------------
+ * 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.
+
 res_pjsip
 ------------------
  * Added "fax_detect_timeout" to endpoint.

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 = "";
@@ -24620,9 +24612,6 @@ 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 */
@@ -24642,7 +24631,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
 		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 */
 	if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite) {