[svn-commits] twilson: trunk r354597 - in /trunk: ./ channels/ channels/sip/ channels/sip/i...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Feb 9 12:14:45 CST 2012
Author: twilson
Date: Thu Feb 9 12:14:39 2012
New Revision: 354597
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=354597
Log:
Add auto_force_rport and auto_comedia NAT options
This patch adds the auto_force_rport and auto_comedia NAT options. It
also converts the nat= setting to a list of comma-separated combinable
options: no, force_rport, comedia, auto_force_rport, and auto_comedia.
nat=yes remains as an undocumented option equal to
"force_rport,comedia". The first instance of 'yes' or 'no' in the list
stops parsing and overrides any previously set options. If an auto_*
option is specified with its non-auto_ counterpart, the auto setting
takes precedence.
This patch builds upon the patch posted to ASTERISK-17860 by JIRA user
pedro-garcia.
(closes issue ASTERISK-17860)
Review: https://reviewboard.asterisk.org/r/1698/
Added:
trunk/channels/sip/utils.c (with props)
Modified:
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/channels/sip/config_parser.c
trunk/channels/sip/include/config_parser.h
trunk/channels/sip/include/sip.h
trunk/channels/sip/include/sip_utils.h
trunk/configs/sip.conf.sample
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Thu Feb 9 12:14:39 2012
@@ -59,6 +59,12 @@
callbackextension options, incoming requests that are matched by address
will be matched to the peer with the matching callbackextension if it is
available.
+ * NAT settings are now a combinable list of options. The equivalent of the
+ deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
+ * Two new NAT options, auto_force_rport and auto_comedia, have been added
+ which set the force_rport and comedia options automatically if Asterisk
+ detects that an incoming SIP request crossed a NAT after being sent by
+ the remote endpoint.
Chan_local changes
------------------
@@ -124,6 +130,12 @@
* MixMonitor will now show IDs associated with the mixmonitor upon creating them
if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
on option to close specific MixMonitors.
+
+ * The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
+ to include information about peers configured with nat=auto_force_rport by
+ returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
+ set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
+ is not enabled.
FAX changes
-----------
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Feb 9 12:14:39 2012
@@ -15276,6 +15276,21 @@
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);
if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri2, XMIT_UNRELIABLE, req->ignore))) {
if (sip_cancel_destroy(p))
@@ -16354,6 +16369,30 @@
ast_sockaddr_set_port(&p->sa,
port != 0 ? port : STANDARD_SIP_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);
+ }
+ }
+
if (sip_debug_test_pvt(p)) {
ast_verbose("Sending to %s (%s)\n",
ast_sockaddr_stringify(sip_real_dst(p)),
@@ -17437,7 +17476,9 @@
snprintf(srch, sizeof(srch), FORMAT2, name,
tmp_host,
peer->host_dynamic ? " D " : " ", /* Dynamic or not? */
- ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */
+ ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ?
+ ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " A " : " a " :
+ ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */
peer->ha ? " A " : " ", /* permit/deny */
tmp_port, status,
peer->description ? peer->description : "",
@@ -17447,7 +17488,9 @@
ast_cli(fd, FORMAT2, name,
tmp_host,
peer->host_dynamic ? " D " : " ", /* Dynamic or not? */
- ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */
+ ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ?
+ ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " A " : " a " :
+ ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : " ", /* NAT=yes? */
peer->ha ? " A " : " ", /* permit/deny */
tmp_port, status,
peer->description ? peer->description : "",
@@ -17462,7 +17505,10 @@
"IPaddress: %s\r\n"
"IPport: %s\r\n"
"Dynamic: %s\r\n"
+ "AutoForcerport: %s\r\n"
"Forcerport: %s\r\n"
+ "AutoComedia: %s\r\n"
+ "Comedia: %s\r\n"
"VideoSupport: %s\r\n"
"TextSupport: %s\r\n"
"ACL: %s\r\n"
@@ -17474,7 +17520,10 @@
ast_sockaddr_isnull(&peer->addr) ? "-none-" : tmp_host,
ast_sockaddr_isnull(&peer->addr) ? "0" : tmp_port,
peer->host_dynamic ? "yes" : "no", /* Dynamic or not? */
+ ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? "yes" : "no",
ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "yes" : "no", /* NAT=yes? */
+ ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ? "yes" : "no",
+ ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "yes" : "no",
ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */
ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no", /* TEXTSUPPORT=yes? */
peer->ha ? "yes" : "no", /* permit/deny */
@@ -18142,7 +18191,8 @@
ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate);
ast_cli(fd, " Expire : %ld\n", ast_sched_when(sched, peer->expire));
ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
- ast_cli(fd, " Force rport : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)));
+ ast_cli(fd, " Force rport : %s\n", force_rport_string(peer->flags));
+ ast_cli(fd, " Symmetric RTP: %s\n", comedia_string(peer->flags));
ast_cli(fd, " ACL : %s\n", AST_CLI_YESNO(peer->ha != NULL));
ast_cli(fd, " DirectMedACL : %s\n", AST_CLI_YESNO(peer->directmediaha != NULL));
ast_cli(fd, " T.38 support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
@@ -18254,7 +18304,12 @@
astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
- astman_append(s, "SIP-Forcerport: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)?"Y":"N"));
+ astman_append(s, "SIP-Forcerport: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ?
+ (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "A" : "a") :
+ (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "Y" : "N"));
+ astman_append(s, "SIP-Comedia: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ?
+ (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "A" : "a") :
+ (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Y" : "N"));
astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
@@ -18866,7 +18921,7 @@
ast_cli(a->fd, " Context: %s\n", sip_cfg.default_context);
ast_cli(a->fd, " Record on feature: %s\n", sip_cfg.default_record_on_feature);
ast_cli(a->fd, " Record off feature: %s\n", sip_cfg.default_record_off_feature);
- ast_cli(a->fd, " Force rport: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_NAT_FORCE_RPORT)));
+ ast_cli(a->fd, " Force rport: %s\n", force_rport_string(global_flags));
ast_cli(a->fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF)));
ast_cli(a->fd, " Qualify: %d\n", default_qualify);
ast_cli(a->fd, " Use ClientCode: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USECLIENTCODE)));
@@ -19237,7 +19292,7 @@
ast_cli(a->fd, " Theoretical Address: %s\n", ast_sockaddr_stringify(&cur->sa));
ast_cli(a->fd, " Received Address: %s\n", ast_sockaddr_stringify(&cur->recv));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
- ast_cli(a->fd, " Force rport: %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[0], SIP_NAT_FORCE_RPORT)));
+ ast_cli(a->fd, " Force rport: %s\n", force_rport_string(cur->flags));
if (ast_sockaddr_isnull(&cur->redirip)) {
ast_cli(a->fd,
" Audio IP: %s (local)\n",
@@ -27692,8 +27747,8 @@
/*!
\brief Handle flag-type options common to configuration of devices - peers
- \param flags array of two struct ast_flags
- \param mask array of two struct ast_flags
+ \param flags array of three struct ast_flags
+ \param mask array of three struct ast_flags
\param v linked list of config variables to process
\returns non-zero if any config options were handled, zero otherwise
*/
@@ -27743,19 +27798,7 @@
ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
}
} else if (!strcasecmp(v->name, "nat")) {
- ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT);
- ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT); /* Default to "force_rport" */
- if (!strcasecmp(v->value, "no")) {
- ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- } else if (!strcasecmp(v->value, "yes")) {
- /* We've already defaulted to force_rport */
- ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
- ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
- } else if (!strcasecmp(v->value, "comedia")) {
- ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
- ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
- ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
- }
+ sip_parse_nat_option(v->value, mask, flags);
} else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
@@ -28969,7 +29012,7 @@
struct sip_peer *peer;
char *cat, *stringp, *context, *oldregcontext;
char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT];
- struct ast_flags dummy[2];
+ struct ast_flags dummy[3];
struct ast_flags config_flags = { reason == CHANNEL_MODULE_LOAD ? 0 : ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) ? 0 : CONFIG_FLAG_FILEUNCHANGED };
int auto_sip_domains = FALSE;
struct ast_sockaddr old_bindaddr = bindaddr;
@@ -29164,7 +29207,7 @@
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
- ast_set_flag(&global_flags[0], SIP_NAT_FORCE_RPORT); /*!< Default to nat=force_rport */
+ ast_set_flag(&global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT); /*!< Default to nat=auto_force_rport */
ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot));
Modified: trunk/channels/sip/config_parser.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/config_parser.c?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/channels/sip/config_parser.c (original)
+++ trunk/channels/sip/config_parser.c Thu Feb 9 12:14:39 2012
@@ -765,11 +765,128 @@
}
+/*! \brief Parse the comma-separated nat= option values */
+void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags)
+{
+ char *parse, *this;
+
+ if (!(parse = ast_strdupa(value))) {
+ return;
+ }
+
+ /* Since we need to completely override the general settings if we are being called
+ * later for a peer, always set the flags for all options on the mask */
+ ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT);
+ ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
+ ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_RPORT);
+ ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
+
+ while ((this = strsep(&parse, ","))) {
+ if (ast_false(this)) {
+ ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
+ ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
+ ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
+ ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
+ break; /* It doesn't make sense to have no + something else */
+ } else if (!strcasecmp(this, "yes")) {
+ ast_log(LOG_WARNING, "nat=yes is deprecated, use nat=force_rport,comedia instead\n");
+ ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
+ ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
+ ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
+ ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
+ break; /* It doesn't make sense to have yes + something else */
+ } else if (!strcasecmp(this, "force_rport") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
+ ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
+ } else if (!strcasecmp(this, "comedia") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
+ ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
+ } else if (!strcasecmp(this, "auto_force_rport")) {
+ ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
+ /* In case someone did something dumb like nat=force_rport,auto_force_rport */
+ ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
+ } else if (!strcasecmp(this, "auto_comedia")) {
+ ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
+ /* In case someone did something dumb like nat=comedia,auto_comedia*/
+ ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
+ }
+ }
+}
+
+#define TEST_FORCE_RPORT 1 << 0
+#define TEST_COMEDIA 1 << 1
+#define TEST_AUTO_FORCE_RPORT 1 << 2
+#define TEST_AUTO_COMEDIA 1 << 3
+static int match_nat_options(int val, struct ast_flags *flags)
+{
+ if ((!ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != !(val & TEST_FORCE_RPORT)) {
+ return 0;
+ }
+ if (!ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) != !(val & TEST_COMEDIA)) {
+ return 0;
+ }
+ if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT) != !(val & TEST_AUTO_FORCE_RPORT)) {
+ return 0;
+ }
+ if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) != !(val & TEST_AUTO_COMEDIA)) {
+ return 0;
+ }
+ return 1;
+}
+
+AST_TEST_DEFINE(sip_parse_nat_test)
+{
+ int i, res = AST_TEST_PASS;
+ struct ast_flags mask[3] = {{0}}, flags[3] = {{0}};
+ struct {
+ const char *str;
+ int i;
+ } options[] = {
+ { "yes", TEST_FORCE_RPORT | TEST_COMEDIA },
+ { "no", 0 },
+ { "force_rport", TEST_FORCE_RPORT },
+ { "comedia", TEST_COMEDIA },
+ { "auto_force_rport", TEST_AUTO_FORCE_RPORT },
+ { "auto_comedia", TEST_AUTO_COMEDIA },
+ { "force_rport,auto_force_rport", TEST_AUTO_FORCE_RPORT },
+ { "auto_force_rport,force_rport", TEST_AUTO_FORCE_RPORT },
+ { "comedia,auto_comedia", TEST_AUTO_COMEDIA },
+ { "auto_comedia,comedia", TEST_AUTO_COMEDIA },
+ { "force_rport,comedia", TEST_FORCE_RPORT | TEST_COMEDIA },
+ { "force_rport,auto_comedia", TEST_FORCE_RPORT | TEST_AUTO_COMEDIA },
+ { "force_rport,yes,no", TEST_FORCE_RPORT | TEST_COMEDIA },
+ { "auto_comedia,no,yes", 0 },
+ };
+
+ switch (cmd) {
+ case TEST_INIT:
+ info->name = "sip_parse_nat_test";
+ info->category = "/channels/chan_sip/";
+ info->summary = "tests sip.conf nat line parsing";
+ info->description =
+ "Tests parsing of various nat line configurations. "
+ "Verifies output matches expected behavior.";
+ return AST_TEST_NOT_RUN;
+ case TEST_EXECUTE:
+ break;
+ }
+
+ for (i = 0; i < ARRAY_LEN(options); i++) {
+ sip_parse_nat_option(options[i].str, mask, flags);
+ if (!match_nat_options(options[i].i, flags)) {
+ ast_test_status_update(test, "Failed nat=%s\n", options[i].str);
+ res = AST_TEST_FAIL;
+ }
+ memset(flags, 0, sizeof(flags));
+ memset(mask, 0, sizeof(mask));
+ }
+
+ return res;
+}
/*! \brief SIP test registration */
void sip_config_parser_register_tests(void)
{
AST_TEST_REGISTER(sip_parse_register_line_test);
AST_TEST_REGISTER(sip_parse_host_line_test);
+ AST_TEST_REGISTER(sip_parse_nat_test);
}
/*! \brief SIP test registration */
@@ -777,5 +894,6 @@
{
AST_TEST_UNREGISTER(sip_parse_register_line_test);
AST_TEST_UNREGISTER(sip_parse_host_line_test);
-}
-
+ AST_TEST_UNREGISTER(sip_parse_nat_test);
+}
+
Modified: trunk/channels/sip/include/config_parser.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/config_parser.h?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/channels/sip/include/config_parser.h (original)
+++ trunk/channels/sip/include/config_parser.h Thu Feb 9 12:14:39 2012
@@ -43,6 +43,18 @@
*/
int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
+/*! \brief Parse the comma-separated nat= option values
+ * \param value The comma-separated value
+ * \param mask An array of ast_flags that will be set by this function
+ * and used as a mask for copying the flags later
+ * \param flags An array of ast_flags that will be set by this function
+ *
+ * \note The nat-related values in both mask and flags are assumed to empty. This function
+ * will treat the first "yes" or "no" value in a list of values as overiding all other values
+ * and will stop parsing. Auto values will override their non-auto counterparts.
+ */
+void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags);
+
/*!
* \brief register config parsing tests
*/
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Thu Feb 9 12:14:39 2012
@@ -363,9 +363,11 @@
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
#define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
+#define SIP_PAGE3_NAT_AUTO_RPORT (1 << 2) /*!< DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected */
+#define SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3) /*!< DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected */
#define SIP_PAGE3_FLAGS_TO_COPY \
- (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32)
+ (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA)
#define CHECK_AUTH_BUF_INITLEN 256
@@ -1062,6 +1064,7 @@
* for incoming calls
*/
unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
+ unsigned short natdetected:1; /*!< Whether we detected a NAT when processing the Via */
char tag[11]; /*!< Our tag for this session */
int timer_t1; /*!< SIP timer T1, ms rtt */
int timer_b; /*!< SIP timer B, ms */
Modified: trunk/channels/sip/include/sip_utils.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip_utils.h?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/channels/sip/include/sip_utils.h (original)
+++ trunk/channels/sip/include/sip_utils.h Thu Feb 9 12:14:39 2012
@@ -80,4 +80,10 @@
*/
const char *hangup_cause2sip(int cause);
+/*! \brief Return a string describing the force_rport value for the given flags */
+const char *force_rport_string(struct ast_flags *flags);
+
+/*! \brief Return a string describing the comedia value for the given flags */
+const char *comedia_string(struct ast_flags *flags);
+
#endif
Added: trunk/channels/sip/utils.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/utils.c?view=auto&rev=354597
==============================================================================
--- trunk/channels/sip/utils.c (added)
+++ trunk/channels/sip/utils.c Thu Feb 9 12:14:39 2012
@@ -1,0 +1,45 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2012, Digium, Inc.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Utility functions for chan_sip
+ *
+ * \author Terry Wilson <twilson at digium.com>
+ */
+
+#include "asterisk.h"
+
+#include "asterisk/utils.h"
+#include "asterisk/cli.h"
+#include "include/sip.h"
+#include "include/sip_utils.h"
+
+const char *force_rport_string(struct ast_flags *flags)
+{
+ if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
+ return ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT) ? "Auto (Yes)" : "Auto (No)";
+ }
+ return AST_CLI_YESNO(ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT));
+}
+
+const char *comedia_string(struct ast_flags *flags)
+{
+ if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
+ return ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Auto (Yes)" : "Auto (No)";
+ }
+ return AST_CLI_YESNO(ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP));
+}
Propchange: trunk/channels/sip/utils.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/channels/sip/utils.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: trunk/channels/sip/utils.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=354597&r1=354596&r2=354597
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Thu Feb 9 12:14:39 2012
@@ -831,17 +831,34 @@
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
-; nat = no ; Use rport if the remote side says to use it.
-; nat = force_rport ; Force rport to always be on. (default)
-; nat = yes ; Force rport to always be on and perform comedia RTP handling.
-; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling.
+; nat = no ; Do no special NAT handling other than RFC3581
+; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
+; nat = comedia ; Send media to the port Asterisk received it from regardless
+; ; of where the SDP says to send it.
+; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
+; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
+;
+; The nat settings can be combined. For example, to set both force_rport and comedia
+; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
+; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
+; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
+; the non-auto option will be ignored.
+;
+; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
+; SIP responses to it via the source IP and port from which the request originated
+; instead of the address/port listed in the top-most Via header. This is useful if a
+; client knows that it is behind a NAT and therefore cannot guess from what address/port
+; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
+; sent. The force_rport setting causes Asterisk to always send responses back to the
+; address/port from which it received requests; even if the other side doesn't support
+; adding the 'rport' parameter.
;
; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
; draft form. This method is used to accomodate endpoints that may be located behind
-; NAT devices, and as such the port number they tell Asterisk to send RTP packets to
-; for their media streams is not actual port number that will be used on the nearer
+; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
+; for their media streams is not the actual address/port that will be used on the nearer
; side of the NAT.
;
; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
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