[Asterisk-code-review] chan sip: Enable Session-Timers for SIP over TCP (and TLS). (asterisk[11])

[Alexander Traud]

Alexander Traud has uploaded a new change for review.

  https://gerrit.asterisk.org/3232

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

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
---
M channels/chan_sip.c
1 file changed, 20 insertions(+), 32 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/32/3232/1

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b309558..1e2a047 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4151,19 +4151,6 @@
 		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 & SIP_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;
-		}
-	}
-
 	if (!(pkt = ast_calloc(1, sizeof(*pkt)))) {
 		return AST_FAILURE;
 	}
@@ -4195,6 +4182,10 @@
 
 	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 & SIP_TRANSPORT_UDP)) {
+		pkt->retrans_stop = 1;
+	}
 
 	/* Schedule retransmission */
 	AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
@@ -23944,6 +23935,7 @@
 	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 = "";
@@ -23972,28 +23964,24 @@
 			ast_channel_hangupcause_set(owner, hangup_sip2cause(resp));
 	}
 
-	if (p->socket.type == SIP_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);
 		}
 
-		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;
-		}
+		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 */

-- 
To view, visit https://gerrit.asterisk.org/3232
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 11
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>