diff options
| -rw-r--r-- | CHANGES | 1 | ||||
| -rw-r--r-- | channels/chan_sip.c | 16 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 7 |
3 files changed, 22 insertions, 2 deletions
@@ -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 diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 2bd9f9523..f270d1f12 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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; diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 42c32607e..8f317305b 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -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. |