[svn-commits] trunk r28215 - in /trunk: channels/chan_sip.c
configs/sip.conf.sample
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu May 18 09:58:00 MST 2006
Author: kpfleming
Date: Thu May 18 11:57:59 2006
New Revision: 28215
URL: http://svn.digium.com/view/asterisk?rev=28215&view=rev
Log:
add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
also, documented the 'canreinvite=update' option in the sample config file
Modified:
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=28215&r1=28214&r2=28215&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu May 18 11:57:59 2006
@@ -636,21 +636,22 @@
#define SIP_NAT_ROUTE (2 << 18) /*!< NAT Only ROUTE */
#define SIP_NAT_ALWAYS (3 << 18) /*!< NAT Both ROUTE and RFC3581 */
/* re-INVITE related settings */
-#define SIP_REINVITE (3 << 20) /*!< two bits used */
+#define SIP_REINVITE (7 << 20) /*!< three bits used */
#define SIP_CAN_REINVITE (1 << 20) /*!< allow peers to be reinvited to send media directly p2p */
-#define SIP_REINVITE_UPDATE (2 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */
+#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< allow media reinvite when new peer is behind NAT */
+#define SIP_REINVITE_UPDATE (4 << 20) /*!< use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings */
-#define SIP_INSECURE_PORT (1 << 22) /*!< don't require matching port for incoming requests */
-#define SIP_INSECURE_INVITE (1 << 23) /*!< don't require authentication for incoming INVITEs */
+#define SIP_INSECURE_PORT (1 << 23) /*!< don't require matching port for incoming requests */
+#define SIP_INSECURE_INVITE (1 << 24) /*!< don't require authentication for incoming INVITEs */
/* Sending PROGRESS in-band settings */
-#define SIP_PROG_INBAND (3 << 24) /*!< three settings, uses two bits */
-#define SIP_PROG_INBAND_NEVER (0 << 24)
-#define SIP_PROG_INBAND_NO (1 << 24)
-#define SIP_PROG_INBAND_YES (2 << 24)
-#define SIP_CALL_ONHOLD (1 << 26) /*!< Call states */
-#define SIP_CALL_LIMIT (1 << 27) /*!< Call limit enforced for this call */
-#define SIP_SENDRPID (1 << 28) /*!< Remote Party-ID Support */
-#define SIP_INC_COUNT (1 << 29) /*!< Did this connection increment the counter of in-use calls? */
+#define SIP_PROG_INBAND (3 << 25) /*!< three settings, uses two bits */
+#define SIP_PROG_INBAND_NEVER (0 << 25)
+#define SIP_PROG_INBAND_NO (1 << 25)
+#define SIP_PROG_INBAND_YES (2 << 25)
+#define SIP_CALL_ONHOLD (1 << 27) /*!< Call states */
+#define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
+#define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
+#define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
#define SIP_FLAGS_TO_COPY \
(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
@@ -12612,10 +12613,24 @@
} else if (!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
- if (!strcasecmp(v->value, "update"))
- ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE);
- else
- ast_set2_flag(&flags[0], ast_true(v->value), SIP_CAN_REINVITE);
+ if (ast_true(v->value)) {
+ ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT);
+ } else if (!ast_false(v->value)) {
+ char buf[64];
+ char *word, *next = buf;
+
+ ast_copy_string(buf, v->value, sizeof(buf));
+ while ((word = strsep(&next, ","))) {
+ if (!strcasecmp(word, "update")) {
+ ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE);
+ } else if (!strcasecmp(word, "nonat")) {
+ ast_set_flag(&flags[0], SIP_CAN_REINVITE);
+ ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT);
+ } else {
+ ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno);
+ }
+ }
+ }
} else if (!strcasecmp(v->name, "insecure")) {
ast_set_flag(&mask[0], SIP_INSECURE_PORT | SIP_INSECURE_INVITE);
ast_clear_flag(&flags[0], SIP_INSECURE_PORT | SIP_INSECURE_INVITE);
@@ -13738,6 +13753,15 @@
ast_mutex_unlock(&p->lock);
return 0;
}
+
+ /* if this peer cannot handle reinvites of the media stream to devices
+ that are known to be behind a NAT, then stop the process now
+ */
+ if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
+ ast_mutex_unlock(&p->lock);
+ return 0;
+ }
+
if (rtp)
changed |= ast_rtp_get_peer(rtp, &p->redirip);
else
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=28215&r1=28214&r2=28215&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Thu May 18 11:57:59 2006
@@ -223,12 +223,20 @@
;canreinvite=yes ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
- ; support this (especially if one of them is
- ; behind a NAT).
+ ; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
- ; behind a NAT, or for some other reason wants
- ; Asterisk to stay in the audio path,
- ; you may want to turn this off
+ ; behind a NAT, or for some other reason wants Asterisk to
+ ; stay in the audio path, you may want to turn this off.
+
+;canreinvite=nonat ; An additional option is to allow media path redirection
+ ; (reinvite) but only when the peer where the media is being
+ ; sent is known to not be behind a NAT (as the RTP core can
+ ; determine it based on the apparent IP address the media
+ ; arrives from).
+
+;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
+ ; instead of INVITE. This can be combined with 'nonat', as
+ ; 'canreinvite=update,nonat'. It implies 'yes'.
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
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