[asterisk-commits] elguero: trunk r385474 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Apr 12 10:06:13 CDT 2013
Author: elguero
Date: Fri Apr 12 10:06:09 2013
New Revision: 385474
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=385474
Log:
Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
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/
........
Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=385474&r1=385473&r2=385474
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Apr 12 10:06:09 2013
@@ -1273,6 +1273,8 @@
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);
@@ -17107,22 +17109,12 @@
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);
- }
- }
-
- ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT);
+
+ 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);
+ }
+
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");
@@ -18151,6 +18143,67 @@
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)
{
@@ -18214,29 +18267,7 @@
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",
@@ -18304,13 +18335,10 @@
* are set on the peer. So we check for that here and set the peer's
* address accordingly.
*/
+ set_peer_nat(p, peer);
+
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);
- }
-
- if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
- ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
}
if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
@@ -30011,6 +30039,22 @@
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. */
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