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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.

Browse Source 
(closes issue #10417)
Reported by: cstadlmann


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Joshua Colp 17 years ago
parent a403fd8ea1
commit 405dfcb54a
3 changed files with 22 additions and 2 deletions
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1
CHANGES
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@ -306,6 +306,7 @@ Changes since Asterisk 1.2:
o canreinvite variable can also now be set to 'nonat'
o rtsavesysname variable added
o JitterBuffer support added
o t38pt_usertpsource variable added
23. skinny.conf
o port variable renamed to bindport
o JitterBuffer support added

16
channels/chan_sip.c
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@ -795,10 +795,11 @@ struct sip_auth {
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< 25: ???? */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< 26: Buggy CISCO MWI fix */
#define SIP_PAGE2_OUTGOING_CALL (1 << 27) /*!< 27: Is this an outgoing call? */
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 28) /*!< 28: 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_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_UDPTL_DESTINATION)
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
@ -5203,6 +5204,16 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
@ -16294,6 +16305,9 @@ 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;

7
configs/sip.conf.sample
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@ -484,7 +484,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; videosupport videosupport
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
; username
; t38pt_usertpsource username
; template
; fromdomain
; regexten
@ -498,6 +498,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; sendrpid
; outboundproxy
; rfc2833compensate
; t38pt_usertpsource
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
@ -662,3 +663,7 @@ 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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