Merged revisions 118647 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines

Merged revisions 118646 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines

Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@118648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -151,6 +151,7 @@ SIP changes
      SIP session.
   * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
      configs/sip.conf.sample for more information on how it is used.
+  * Added t38pt_usertpsource option. See sip.conf.sample for details.
 
 IAX2 changes
 ------------

channels/chan_sip.c

@@ -987,11 +987,12 @@ struct sip_auth {
 #define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)	/*!< DP: Compensate for buggy RFC2833 implementations */
 #define SIP_PAGE2_BUGGY_MWI		(1 << 26)	/*!< DP: Buggy CISCO MWI fix */
 #define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
+#define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
 	SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
-	SIP_PAGE2_TEXTSUPPORT )
+	SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION)
 
 /*@}*/ 
 
@@ -6556,6 +6557,16 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	if (p->udptl) {
 		if (udptlportno > 0) {
 			sin.sin_port = htons(udptlportno);
+			if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
+				struct sockaddr_in peer;
+				ast_rtp_get_peer(p->rtp, &peer);
+				if (peer.sin_addr.s_addr) {
+					memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(&sin.sin_addr));
+					if (debug) {
+						ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
+					}
+				}
+			}
 			ast_udptl_set_peer(p->udptl, &sin);
 			if (debug)
 				ast_debug(1, "Peer T.38 UDPTL is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
@@ -19286,6 +19297,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 	} else if (!strcasecmp(v->name, "buggymwi")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
+	} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
 	} else
 		res = 0;
 

configs/sip.conf.sample

@@ -691,14 +691,14 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ; videosupport		      videosupport
 ; maxcallbitrate	      maxcallbitrate
 ; rfc2833compensate           mailbox
-; session-timers             busylevel
+; session-timers              busylevel
 ; session-expires            
-; session-minse              template
+; session-minse               template
-; session-refresher          fromdomain
+; session-refresher           fromdomain
-;                            regexten
+; t38pt_usertpsource          regexten
-;                            fromuser
+;                             fromuser
-;                            host
+;                             host
-;                            port
+;                             port
 ;                             qualify
 ;                             defaultip
 ;                             defaultuser
@@ -716,7 +716,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;                             timert1
 ;                             timerb
 ;                             qualifyfreq
-
+;                             t38pt_usertpsource
 
 ;[sip_proxy]
 ; For incoming calls only. Example: FWD (Free World Dialup)
@@ -935,3 +935,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;host=dynamic
 ;rfc2833compensate=yes		; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
 				; You must have this turned on or DTMF reception will work improperly.
+;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
+                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
+                                ; external IP address of the remote device. If port forwarding is done at the client side
+                                ; then UDPTL will flow to the remote device.