Merged revisions 118646 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
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/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.1
Joshua Colp 17 years ago
parent bf13b4df4e
commit e4d1b39bd8

@ -97,6 +97,7 @@ SIP Changes
the register line.
* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
lost packets.
* Added t38pt_usertpsource option. See sip.conf.sample for details.
IAX Changes
-----------

@ -1010,11 +1010,12 @@ 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)
/*@}*/
@ -6889,6 +6890,16 @@ 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));
@ -20455,6 +20466,9 @@ 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;

@ -710,14 +710,14 @@ 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-refresher fromdomain
; regexten
; fromuser
; host
; port
; session-minse template
; session-refresher fromdomain
; t38pt_usertpsource regexten
; fromuser
; host
; port
; qualify
; defaultip
; defaultuser
@ -735,7 +735,7 @@ 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)
@ -956,3 +956,7 @@ 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.

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