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-rw-r--r--channels/chan_sip.c213
1 files changed, 103 insertions, 110 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8f3296adf..9cc6252fd 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);
@@ -7665,7 +7656,8 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
change_t38_state(p, T38_ENABLED);
transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
- } else if (p->t38.state != T38_ENABLED) {
+ } else if ((p->t38.state != T38_ENABLED) || ((p->t38.state == T38_ENABLED) &&
+ (parameters->request_response == AST_T38_REQUEST_NEGOTIATE))) {
p->t38.our_parms = *parameters;
ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
change_t38_state(p, T38_LOCAL_REINVITE);
@@ -8606,29 +8598,31 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_CNG)) {
if (strcmp(ast_channel_exten(ast), "fax")) {
const char *target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));
- /* We need to unlock 'ast' here because
+ /*
+ * We need to unlock 'ast' here because
* ast_exists_extension has the potential to start and
* stop an autoservice on the channel. Such action is
* prone to deadlock if the channel is locked.
+ *
+ * ast_async_goto() has its own restriction on not holding
+ * the channel lock.
*/
sip_pvt_unlock(p);
ast_channel_unlock(ast);
+ ast_frfree(fr);
+ fr = &ast_null_frame;
if (ast_exists_extension(ast, target_context, "fax", 1,
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, NULL))) {
- ast_channel_lock(ast);
- sip_pvt_lock(p);
ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n", ast_channel_name(ast));
pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
if (ast_async_goto(ast, target_context, "fax", 1)) {
ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(ast), target_context);
}
- ast_frfree(fr);
- fr = &ast_null_frame;
} else {
- ast_channel_lock(ast);
- sip_pvt_lock(p);
ast_log(LOG_NOTICE, "FAX CNG detected but no fax extension\n");
}
+ ast_channel_lock(ast);
+ sip_pvt_lock(p);
}
}
@@ -11334,25 +11328,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
}
- if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) {
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 32000) {
- ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- } else {
- found = TRUE;
- }
- }
- } else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) {
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 48000) {
- ast_log(LOG_WARNING, "Got Siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
- } else {
- found = TRUE;
- }
- }
- } else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
+ if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 64000) {
ast_log(LOG_WARNING, "Got G.719 offer at %u bps, but only 64000 bps supported; ignoring.\n", bit_rate);
@@ -13011,12 +12987,6 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
} else if (ast_format_cmp(format, ast_format_g723) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we don't support VAD (G.723.1 annex A) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
- } else if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) {
- /* Indicate that we only expect 32Kbps */
- ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
- } else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) {
- /* Indicate that we only expect 48Kbps */
- ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
} else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we only expect 64Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
@@ -14156,9 +14126,10 @@ static void build_contact(struct sip_pvt *p, struct sip_request *req, int incomi
/*! \brief Initiate new SIP request to peer/user */
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, const char * const explicit_uri)
{
- struct ast_str *invite = ast_str_alloca(256);
- char from[256];
- char to[256];
+#define SIPHEADER 256
+ struct ast_str *invite = ast_str_create(SIPHEADER);
+ struct ast_str *from = ast_str_create(SIPHEADER);
+ struct ast_str *to = ast_str_create(SIPHEADER);
char tmp_n[SIPBUFSIZE/2]; /* build a local copy of 'n' if needed */
char tmp_l[SIPBUFSIZE/2]; /* build a local copy of 'l' if needed */
const char *l = NULL; /* XXX what is this, exactly ? */
@@ -14260,34 +14231,40 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
ourport = (p->fromdomainport && (p->fromdomainport != STANDARD_SIP_PORT)) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
if (!sip_standard_port(p->socket.type, ourport)) {
- ret = snprintf(from, sizeof(from), "<sip:%s@%s:%d>;tag=%s", tmp_l, d, ourport, p->tag);
+ ret = ast_str_set(&from, 0, "<sip:%s@%s:%d>;tag=%s", tmp_l, d, ourport, p->tag);
} else {
- ret = snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", tmp_l, d, p->tag);
+ ret = ast_str_set(&from, 0, "<sip:%s@%s>;tag=%s", tmp_l, d, p->tag);
}
- if (ret < 0 || ret >= sizeof(from)) { /* a return value of size or more means that the output was truncated */
+ if (ret == AST_DYNSTR_BUILD_FAILED) {
/* We don't have an escape path from here... */
ast_log(LOG_ERROR, "The From header was truncated in call '%s'. This call setup will fail.\n", p->callid);
+ /* Make sure that the field contains something non-broken.
+ See https://issues.asterisk.org/jira/browse/ASTERISK-26069
+ */
+ ast_str_set(&from, 3, "<>");
+
}
/* If a caller id name was specified, prefix a display name, if there is enough room. */
if (cid_has_name || !cid_has_num) {
- size_t written = strlen(from);
- ssize_t left = sizeof(from) - written - 4; /* '"" \0' */
- if (left > 0) {
- size_t name_len;
- if (sip_cfg.pedanticsipchecking) {
- ast_escape_quoted(n, tmp_n, MIN(left + 1, sizeof(tmp_n)));
- n = tmp_n;
- }
- name_len = strlen(n);
- if (left < name_len) {
- name_len = left;
- }
- memmove(from + name_len + 3, from, written + 1);
- from[0] = '"';
- memcpy(from + 1, n, name_len);
- from[name_len + 1] = '"';
- from[name_len + 2] = ' ';
+ size_t written = ast_str_strlen(from);
+ size_t name_len;
+ if (sip_cfg.pedanticsipchecking) {
+ ast_escape_quoted(n, tmp_n, sizeof(tmp_n));
+ n = tmp_n;
+ }
+ name_len = strlen(n);
+ ret = ast_str_make_space(&from, name_len + written + 4);
+
+ if (ret == 0) {
+ /* needed again, as ast_str_make_space coud've changed the pointer */
+ char *from_buf = ast_str_buffer(from);
+
+ memmove(from_buf + name_len + 3, from_buf, written + 1);
+ from_buf[0] = '"';
+ memcpy(from_buf + 1, n, name_len);
+ from_buf[name_len + 1] = '"';
+ from_buf[name_len + 2] = ' ';
}
}
@@ -14330,24 +14307,28 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
/*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
if (!strchr(p->todnid, '@')) {
/* We have no domain in the dnid */
- ret = snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
+ ret = ast_str_set(&to, 0, "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
} else {
- ret = snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
+ ret = ast_str_set(&to, 0, "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
}
} else {
if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
- ret = snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
+ ret = ast_str_set(&to, 0, "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
} else if (p->options && p->options->vxml_url) {
/* If there is a VXML URL append it to the SIP URL */
- ret = snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
+ ret = ast_str_set(&to, 0, "<%s>;%s", p->uri, p->options->vxml_url);
} else {
- ret = snprintf(to, sizeof(to), "<%s>", p->uri);
+ ret = ast_str_set(&to, 0, "<%s>", p->uri);
}
}
- if (ret < 0 || ret >= sizeof(to)) { /* a return value of size or more means that the output was truncated */
+ if (ret == AST_DYNSTR_BUILD_FAILED) {
/* We don't have an escape path from here... */
ast_log(LOG_ERROR, "The To header was truncated in call '%s'. This call setup will fail.\n", p->callid);
+ /* Make sure that the field contains something non-broken.
+ See https://issues.asterisk.org/jira/browse/ASTERISK-26069
+ */
+ ast_str_set(&to, 3, "<>");
}
init_req(req, sipmethod, p->uri);
@@ -14362,8 +14343,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
*/
add_route(req, &p->route, 0);
- add_header(req, "From", from);
- add_header(req, "To", to);
+ add_header(req, "From", ast_str_buffer(from));
+ add_header(req, "To", ast_str_buffer(to));
ast_string_field_set(p, exten, l);
build_contact(p, req, 0);
add_header(req, "Contact", p->our_contact);
@@ -14372,6 +14353,10 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
if (!ast_strlen_zero(global_useragent)) {
add_header(req, "User-Agent", global_useragent);
}
+
+ ast_free(from);
+ ast_free(to);
+ ast_free(invite);
}
/*! \brief Add "Diversion" header to outgoing message
@@ -17196,10 +17181,8 @@ static void mwi_event_cb(void *userdata, struct stasis_subscription *sub, struct
struct sip_peer *peer = sip_find_peer(peer_name, NULL, TRUE, FINDALLDEVICES, FALSE, 0);
if (stasis_subscription_final_message(sub, msg)) {
- if (peer) {
- /* configuration reloaded */
- return;
- }
+ /* peer can be non-NULL during reload. */
+ ao2_cleanup(peer);
ast_free(peer_name);
return;
}
@@ -18353,7 +18336,7 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
static int get_sip_pvt_from_replaces(const char *callid, const char *totag,
const char *fromtag, struct sip_pvt **out_pvt, struct ast_channel **out_chan)
{
- struct sip_pvt *sip_pvt_ptr;
+ RAII_VAR(struct sip_pvt *, sip_pvt_ptr, NULL, ao2_cleanup);
struct sip_pvt tmp_dialog = {
.callid = callid,
};
@@ -18428,6 +18411,9 @@ static int get_sip_pvt_from_replaces(const char *callid, const char *totag,
}
}
+ /* If we're here sip_pvt_ptr has been copied to *out_pvt, prevent RAII_VAR cleanup */
+ sip_pvt_ptr = NULL;
+
return 0;
}
@@ -21287,15 +21273,13 @@ static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
}
/*! \brief Callback for show_chanstats */
-static int show_chanstats_cb(void *__cur, void *__arg, int flags)
+static int show_chanstats_cb(struct sip_pvt *cur, struct __show_chan_arg *arg)
{
#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define FORMAT "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"
- struct sip_pvt *cur = __cur;
struct ast_rtp_instance_stats stats;
char durbuf[10];
struct ast_channel *c;
- struct __show_chan_arg *arg = __arg;
int fd = arg->fd;
sip_pvt_lock(cur);
@@ -21355,6 +21339,8 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
+ struct sip_pvt *cur;
+ struct ao2_iterator i;
switch (cmd) {
case CLI_INIT:
@@ -21372,8 +21358,14 @@ static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_
return CLI_SHOWUSAGE;
ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "Jitter", "Send: Pack", "Lost", "Jitter");
+
/* iterate on the container and invoke the callback on each item */
- ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
+ i = ao2_iterator_init(dialogs, 0);
+ for (; (cur = ao2_iterator_next(&i)); ao2_ref(cur, -1)) {
+ show_chanstats_cb(cur, &arg);
+ }
+ ao2_iterator_destroy(&i);
+
ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : "");
return CLI_SUCCESS;
}
@@ -21689,10 +21681,8 @@ static const struct cfsubscription_types *find_subscription_type(enum subscripti
#define FORMAT "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s %-10.10s\n"
/*! \brief callback for show channel|subscription */
-static int show_channels_cb(void *__cur, void *__arg, int flags)
+static int show_channels_cb(struct sip_pvt *cur, struct __show_chan_arg *arg)
{
- struct sip_pvt *cur = __cur;
- struct __show_chan_arg *arg = __arg;
const struct ast_sockaddr *dst;
sip_pvt_lock(cur);
@@ -21744,7 +21734,8 @@ static int show_channels_cb(void *__cur, void *__arg, int flags)
static char *sip_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
-
+ struct sip_pvt *cur;
+ struct ao2_iterator i;
if (cmd == CLI_INIT) {
e->command = "sip show {channels|subscriptions}";
@@ -21766,7 +21757,11 @@ static char *sip_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(arg.fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type", "Mailbox", "Expiry");
/* iterate on the container and invoke the callback on each item */
- ao2_t_callback(dialogs, OBJ_NODATA, show_channels_cb, &arg, "callback to show channels");
+ i = ao2_iterator_init(dialogs, 0);
+ for (; (cur = ao2_iterator_next(&i)); ao2_ref(cur, -1)) {
+ show_channels_cb(cur, &arg);
+ }
+ ao2_iterator_destroy(&i);
/* print summary information */
ast_cli(arg.fd, "%d active SIP %s%s\n", arg.numchans,
@@ -24590,6 +24585,7 @@ 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 = "";
@@ -24618,28 +24614,24 @@ 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 */
@@ -31041,6 +31033,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
return NULL;
}
if (!(peer->endpoint = ast_endpoint_create("SIP", name))) {
+ ao2_t_ref(peer, -1, "failed to allocate endpoint, drop peer");
return NULL;
}
if (!(peer->caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-22 15:06:06 +0000