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Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."
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8
CHANGES
8
CHANGES
@@ -181,6 +181,14 @@ chan_sip
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NOTE: This is again separated by an exclamation mark, so the To: header may
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not contain one of those.
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* Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
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Previously Asterisk dropped calls only with UDP transports. However with
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longer international calls via TCP, the SIP channel might break, because
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all hops on the Internet route must stay online (have not a single power
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outage, for example). Therefore with Session-Timers enabled (which are
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enabled at default), you might see additional dropped calls. Consequently
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please, consider to go for session-timers=refuse in your sip.conf.
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chan_pjsip
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------------------
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* New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
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@@ -4205,19 +4205,6 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, in
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p->pendinginvite = seqno;
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}
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/* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
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/* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
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/*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
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if (!(p->socket.type & AST_TRANSPORT_UDP)) {
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xmitres = __sip_xmit(p, data); /* Send packet */
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if (xmitres == XMIT_ERROR) { /* Serious network trouble, no need to try again */
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append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
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return AST_FAILURE;
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} else {
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return AST_SUCCESS;
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}
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}
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pkt = ao2_alloc_options(sizeof(*pkt), sip_pkt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
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if (!pkt) {
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return AST_FAILURE;
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@@ -4254,6 +4241,10 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, in
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pkt->time_sent = ast_tvnow(); /* time packet was sent */
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pkt->retrans_stop_time = 64 * (pkt->timer_t1 ? pkt->timer_t1 : DEFAULT_TIMER_T1); /* time in ms after pkt->time_sent to stop retransmission */
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if (!(p->socket.type & AST_TRANSPORT_UDP)) {
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pkt->retrans_stop = 1;
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}
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/* Schedule retransmission */
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ao2_t_ref(pkt, +1, "Schedule packet retransmission");
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pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
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@@ -24654,6 +24645,7 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
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char *c_copy = ast_strdupa(c);
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/* Skip the Cseq and its subsequent spaces */
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const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
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int ack_res = FALSE;
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if (!msg)
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msg = "";
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@@ -24668,28 +24660,24 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
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}
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}
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if (p->socket.type == AST_TRANSPORT_UDP) {
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int ack_res = FALSE;
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/* Acknowledge whatever it is destined for */
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if ((resp >= 100) && (resp <= 199)) {
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/* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
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if (sipmethod == SIP_INVITE) {
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ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
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}
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} else {
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ack_res = __sip_ack(p, seqno, 0, sipmethod);
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}
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/* Acknowledge whatever it is destined for */
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if ((resp >= 100) && (resp <= 199)) {
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/* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
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if (sipmethod == SIP_INVITE) {
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ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
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}
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} else {
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ack_res = __sip_ack(p, seqno, 0, sipmethod);
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if (ack_res == FALSE) {
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/* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
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if (sipmethod == SIP_INVITE && resp >= 200) {
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
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}
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if (ack_res == FALSE) {
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/* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
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if (sipmethod == SIP_INVITE && resp >= 200) {
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
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}
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append_history(p, "Ignore", "Ignoring this retransmit\n");
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return;
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}
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append_history(p, "Ignore", "Ignoring this retransmit\n");
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return;
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}
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/* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
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