Improve handling network errors on transmission to hosts that don't reply or are unreachable
With this code, the call will fail as soon as we get a network error. This may happen on first xmit or a later one, so the retransmit code handles this too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64142 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej
parent
ade74cbe2e
commit
a055b8f5c6
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@ -1915,6 +1915,15 @@ static int __sip_xmit(struct sip_pvt *p, char *data, int len)
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const struct sockaddr_in *dst = sip_real_dst(p);
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res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
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if (res == -1) {
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switch (errno) {
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case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
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case EHOSTUNREACH: /* Host can't be reached */
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case ENETDOWN: /* Inteface down */
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case ENETUNREACH: /* Network failure */
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res = -2; /* Don't bother with trying to transmit again */
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}
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}
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if (res != len)
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ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
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return res;
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@ -2009,6 +2018,7 @@ static int retrans_pkt(void *data)
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{
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struct sip_pkt *pkt = data, *prev, *cur = NULL;
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int reschedule = DEFAULT_RETRANS;
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int xmitres = 0;
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/* Lock channel PVT */
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sip_pvt_lock(pkt->owner);
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@ -2048,19 +2058,27 @@ static int retrans_pkt(void *data)
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}
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append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
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__sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
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xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
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sip_pvt_unlock(pkt->owner);
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return reschedule;
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if (xmitres == -2) {
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ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner->callid);
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} else {
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return reschedule;
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}
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}
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/* Too many retries */
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if (pkt->owner && pkt->method != SIP_OPTIONS) {
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if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
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if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
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ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
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} else {
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if ((pkt->method == SIP_OPTIONS) && sipdebug)
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} else if (pkt->method == SIP_OPTIONS && sipdebug) {
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ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
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}
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append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
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}
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if (xmitres == -2) {
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ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission on Call ID %s\n", pkt->owner->callid);
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append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
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} else
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append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
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pkt->retransid = -1;
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@ -2105,6 +2123,7 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int res
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{
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struct sip_pkt *pkt;
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int siptimer_a = DEFAULT_RETRANS;
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int xmitres = 0;
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if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
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return AST_FAILURE;
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@ -2129,13 +2148,20 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int res
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ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid);
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pkt->next = p->packets;
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p->packets = pkt;
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__sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
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if (sipmethod == SIP_INVITE) {
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/* Note this is a pending invite */
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p->pendinginvite = seqno;
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}
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return AST_SUCCESS;
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xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
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if (xmitres == -2) { /* Serious network trouble, no need to try again */
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append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
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ast_sched_del(sched, pkt->retransid); /* No more retransmission */
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pkt->retransid = -1;
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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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/*! \brief Kill a SIP dialog (called by scheduler) */
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@ -2333,7 +2359,7 @@ static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmitty
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(tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
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}
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res = (reliable) ?
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__sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
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__sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
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__sip_xmit(p, req->data, req->len);
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if (res > 0)
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return 0;
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@ -3163,14 +3189,19 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
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res = -1;
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} else {
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int xmitres;
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p->t38.jointcapability = p->t38.capability;
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if (option_debug > 1)
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ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
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transmit_invite(p, SIP_INVITE, 1, 2);
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p->invitestate = INV_CALLING;
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xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
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if (xmitres != -2) {
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p->invitestate = INV_CALLING;
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/* Initialize auto-congest time */
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p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
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/* Initialize auto-congest time */
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p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
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} else
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res = -1;
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}
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return res;
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@ -6991,7 +7022,9 @@ static void copy_request(struct sip_request *dst, const struct sip_request *src)
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dst->rlPart2 += offset;
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}
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/*! \brief Used for 200 OK and 183 early media */
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/*! \brief Used for 200 OK and 183 early media
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\return Will return -2 for network errors.
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*/
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static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
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{
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struct sip_request resp;
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@ -8090,7 +8123,9 @@ static int transmit_info_with_vidupdate(struct sip_pvt *p)
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return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
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}
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/*! \brief Transmit generic SIP request */
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/*! \brief Transmit generic SIP request
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returns -2 if transmit failed with a critical error (don't retry)
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*/
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static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
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{
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struct sip_request resp;
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@ -12495,6 +12530,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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{
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int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
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int res = 0;
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int xmitres = 0;
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int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
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struct ast_channel *bridgepeer = NULL;
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@ -12678,14 +12714,14 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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}
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/* If I understand this right, the branch is different for a non-200 ACK only */
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p->invitestate = INV_TERMINATED;
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
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check_pendings(p);
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break;
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case 407: /* Proxy authentication */
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case 401: /* Www auth */
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/* First we ACK */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->options)
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p->options->auth_type = resp;
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@ -12706,7 +12742,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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case 403: /* Forbidden */
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/* First we ACK */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
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if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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@ -12715,7 +12751,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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break;
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case 404: /* Not found */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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sip_alreadygone(p);
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@ -12724,7 +12760,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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case 481: /* Call leg does not exist */
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/* Could be REFER caused INVITE with replaces */
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ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner)
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
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@ -12733,14 +12769,14 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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/* We have sent CANCEL on an outbound INVITE
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This transaction is already scheduled to be killed by sip_hangup().
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*/
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transmit_request(p, SIP_ACK, seqno, 0, 0);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
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ast_queue_hangup(p->owner);
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else if (!ast_test_flag(req, SIP_PKT_IGNORE))
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update_call_counter(p, DEC_CALL_LIMIT);
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break;
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case 488: /* Not acceptable here */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (reinvite && p->udptl) {
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/* If this is a T.38 call, we should go back to
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audio. If this is an audio call - something went
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@ -12770,18 +12806,20 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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/* we really should have to wait a while, then retransmit */
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/* We should support the retry-after at some point */
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/* At this point, we treat this as a congestion */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
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break;
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case 501: /* Not implemented */
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transmit_request(p, SIP_ACK, seqno, 0, 0);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner)
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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break;
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}
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if (xmitres == -2)
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ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
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}
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/* \brief Handle SIP response in REFER transaction
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@ -13934,6 +13972,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
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/* Answer the incoming call and set channel to UP state */
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transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
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ast_setstate(c, AST_STATE_UP);
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/* Stop music on hold and other generators */
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@ -16075,6 +16114,7 @@ static int sip_poke_noanswer(void *data)
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static int sip_poke_peer(struct sip_peer *peer)
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{
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struct sip_pvt *p;
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int xmitres = 0;
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if (!peer->maxms || !peer->addr.sin_addr.s_addr) {
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/* IF we have no IP, or this isn't to be monitored, return
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@ -16120,12 +16160,15 @@ static int sip_poke_peer(struct sip_peer *peer)
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ast_set_flag(&p->flags[0], SIP_OUTGOING);
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#ifdef VOCAL_DATA_HACK
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ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
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transmit_invite(p, SIP_INVITE, 0, 2);
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xmitres = transmit_invite(p, SIP_INVITE, 0, 2);
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#else
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transmit_invite(p, SIP_OPTIONS, 0, 2);
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xmitres = transmit_invite(p, SIP_OPTIONS, 0, 2);
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#endif
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gettimeofday(&peer->ps, NULL);
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peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer);
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if (xmitres == -2)
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sip_poke_noanswer(peer); /* Immediately unreachable, network problems */
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else
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peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer);
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return 0;
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}
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