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Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX

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When we reload Asterisk or chan_sip, the flags force_rport and comedia that are
turned on and off when using the auto_force_rport and auto_comedia nat settings
go back to the default setting off.  These flags are turned on when needed or
off when not needed at the time that a peer registers, re-registers or initiates
a call.  This would apply even when only the default global setting
"nat=auto_force_rport" is being used, which in this case would only affect the
force_rport flag.

Everything is good except for the following:  The nat setting is set to
auto_force_rport and auto_comedia.  We reload Asterisk and the peer's
registration has not expired.  We load in the settings for the peer which turns
force_rport and comedia back to off.  Since the peer has not re-registered or
placed a call yet, those flags remain off.  We then initiate a call to the peer
from the PBX.  The force_rport and comedia flags stay off.  If NAT is involved,
we end up with one-way audio since we never checked to see if the peer is behind
NAT or not.

This patch does the following:

* Moves the checking of whether a peer is behind NAT into its own function

* Create a function to set the peer's NAT flags if they are using the auto_* NAT
  settings

* Adds calls in sip_request_call() to these new functions in order to setup the
  dialog according to the peer's settings

(closes issue ASTERISK-21374)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
    asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2421/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Michael L. Young
2013-04-12 15:01:39 +00:00
parent 82e70b2128
commit f07cccecfd
1 changed files with 87 additions and 43 deletions
Download Patch File Download Diff File Expand all files Collapse all files

130
channels/chan_sip.c
View File

@@ -1447,6 +1447,8 @@ static int sip_notify_alloc(struct sip_pvt *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer);
static void check_for_nat(const struct ast_sockaddr *them, struct sip_pvt *p);
/*--- Device monitoring and Device/extension state/event handling */
static int extensionstate_update(const char *context, const char *exten, struct state_notify_data *data, struct sip_pvt *p, int force);
@@ -16862,22 +16864,12 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sock
ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
res = AUTH_PEER_NOT_DYNAMIC;
} else {
if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
if (p->natdetected) {
ast_set_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
} else {
ast_clear_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
}
}
if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
if (p->natdetected) {
ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
} else {
ast_clear_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
set_peer_nat(p, peer);
if (p->natdetected && ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT);
}
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT);
if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri2, XMIT_UNRELIABLE))) {
if (sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
@@ -17889,6 +17881,67 @@ static int get_also_info(struct sip_pvt *p, struct sip_request *oreq)
return -1;
}
/*! \brief Set the peers nat flags if they are using auto_* settings */
static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer)
{
if (!p || !peer) {
return;
}
if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
if (p->natdetected) {
ast_set_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
} else {
ast_clear_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
}
}
if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
if (p->natdetected) {
ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
} else {
ast_clear_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
}
}
/*! \brief Check and see if the requesting UA is likely to be behind a NAT.
*
* If the requesting NAT is behind NAT, set the * natdetected flag so that
* later, peers with nat=auto_* can use the value. Also, set the flags so
* that Asterisk responds identically whether or not a peer exists so as
* not to leak peer name information.
*/
static void check_for_nat(const struct ast_sockaddr *addr, struct sip_pvt *p)
{
if (!addr || !p) {
return;
}
if (ast_sockaddr_cmp(addr, &p->recv)) {
char *tmp_str = ast_strdupa(ast_sockaddr_stringify(addr));
ast_debug(3, "NAT detected for %s / %s\n", tmp_str, ast_sockaddr_stringify(&p->recv));
p->natdetected = 1;
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_set_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
}
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_set_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
} else {
p->natdetected = 0;
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_clear_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
}
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_clear_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
}
}
/*! \brief check Via: header for hostname, port and rport request/answer */
static void check_via(struct sip_pvt *p, const struct sip_request *req)
{
@@ -17952,29 +18005,7 @@ static void check_via(struct sip_pvt *p, const struct sip_request *req)
ast_sockaddr_set_port(&p->sa, port);
/* Check and see if the requesting UA is likely to be behind a NAT. If they are, set the
* natdetected flag so that later, peers with nat=auto_* can use the value. Also
* set the flags so that Asterisk responds identically whether or not a peer exists
* so as not to leak peer name information. */
if (ast_sockaddr_cmp(&tmp, &p->recv)) {
char *tmp_str = ast_strdupa(ast_sockaddr_stringify(&tmp));
ast_debug(3, "NAT detected for %s / %s\n", tmp_str, ast_sockaddr_stringify(&p->recv));
p->natdetected = 1;
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_set_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
}
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_set_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
} else {
p->natdetected = 0;
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_clear_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
}
if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_clear_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
}
check_for_nat(&tmp, p);
if (sip_debug_test_pvt(p)) {
ast_verbose("Sending to %s (%s)\n",
@@ -18042,13 +18073,10 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
* are set on the peer. So we check for that here and set the peer's
* address accordingly.
*/
if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_set_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
ast_sockaddr_copy(&peer->addr, &p->recv);
}
set_peer_nat(p, peer);
if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_sockaddr_copy(&peer->addr, &p->recv);
}
if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
@@ -29668,6 +29696,22 @@ static struct ast_channel *sip_request_call(const char *type, struct ast_format_
ast_string_field_set(p, peername, ext);
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
/* When chan_sip is first loaded, we may have a peer entry but it hasn't re-registered yet.
If the peer hasn't re-registered, we have not checked for NAT yet. With the new
auto_* settings, we need to check for NAT so we do not have one-way audio. */
check_for_nat(&p->ourip, p);
set_peer_nat(p, p->relatedpeer);
if (p->natdetected && ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_copy_flags(&p->flags[0], &p->relatedpeer->flags[0], SIP_NAT_FORCE_RPORT);
}
if (p->natdetected && ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_copy_flags(&p->flags[1], &p->relatedpeer->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
do_setnat(p);
build_via(p);
/* Change the dialog callid. */