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-rw-r--r--channel.c4
-rw-r--r--channels/chan_sip.c886
2 files changed, 873 insertions, 17 deletions
diff --git a/channel.c b/channel.c
index 398bf59ab..30a1ad073 100644
--- a/channel.c
+++ b/channel.c
@@ -2383,6 +2383,10 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
res = (chan->tech->write_video == NULL) ? 0 :
chan->tech->write_video(chan, fr);
break;
+ case AST_FRAME_MODEM:
+ res = (chan->tech->write == NULL) ? 0 :
+ chan->tech->write(chan, fr);
+ break;
case AST_FRAME_VOICE:
if (chan->tech->write == NULL)
break; /*! \todo XXX should return 0 maybe ? */
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5f600814b..49b8aac87 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -118,6 +118,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
+#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
@@ -718,9 +719,13 @@ struct sip_auth {
#define SIP_PAGE2_ALLOWOVERLAP (1 << 11) /*!< Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 12) /*!< Only issue MWI notification if subscribed to */
#define SIP_PAGE2_INC_RINGING (1 << 13) /*!< Did this connection increment the counter of in-use calls? */
+#define SIP_PAGE2_T38SUPPORT (7 << 14) /*!< T38 Fax Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 14) /*!< 14: T38 Fax Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_RTP (2 << 14) /*!< 15: T38 Fax Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_TCP (4 << 14) /*!< 16: T38 Fax Passthrough Support */
#define SIP_PAGE2_FLAGS_TO_COPY \
- (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT)
+ (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT)
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
@@ -729,10 +734,55 @@ struct sip_auth {
#define SIP_PKT_IGNORE_RESP (1 << 3) /*!< Resp ignore - ??? */
#define SIP_PKT_IGNORE_REQ (1 << 4) /*!< Req ignore - ??? */
+/* T.38 set of flags */
+#define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
+#define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
+#define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
+/* Rate management */
+#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
+#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
+/* UDP Error correction */
+#define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
+#define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
+#define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
+/* T38 Spec version */
+#define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
+#define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
+#define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
+/* Maximum Fax Rate */
+#define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
+#define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
+#define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
+#define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
+#define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
+#define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
+
+/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
+static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
+
#define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
#define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
#define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
+/*! \brief T38 Sates for a call */
+enum t38state {
+ T38_DISABLED = 0, /*! Not enabled */
+ T38_LOCAL_DIRECT, /*! Offered from local */
+ T38_LOCAL_REINVITE, /*! Offered from local - REINVITE */
+ T38_PEER_DIRECT, /*! Offered from peer */
+ T38_PEER_REINVITE, /*! Offered from peer - REINVITE */
+ T38_ENABLED /*! Negotiated (enabled) */
+};
+
+/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
+struct t38properties {
+ struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
+ int capability; /*!< Our T38 capability */
+ int peercapability; /*!< Peers T38 capability */
+ int jointcapability; /*!< Supported T38 capability at both ends */
+ enum t38state state; /*!< T.38 state */
+};
+
/*! \brief Parameters to know status of transfer */
enum referstatus {
REFER_IDLE, /*!< No REFER is in progress */
@@ -837,6 +887,9 @@ static struct sip_pvt {
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
+ struct t38properties t38; /*!< T38 settings */
+ struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
+ struct ast_udptl *udptl; /*!< T.38 UDPTL session */
int callingpres; /*!< Calling presentation */
int authtries; /*!< Times we've tried to authenticate */
int expiry; /*!< How long we take to expire */
@@ -1409,8 +1462,14 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan);
static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan);
static int sip_get_codec(struct ast_channel *chan);
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p);
+static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
+/*------ T38 Support --------- */
+static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
+static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
+static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p);
+static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
+static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
@@ -1452,6 +1511,13 @@ static struct ast_rtp_protocol sip_rtp = {
get_codec: sip_get_codec,
};
+/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
+static struct ast_udptl_protocol sip_udptl = {
+ type: "SIP",
+ get_udptl_info: sip_get_udptl_peer,
+ set_udptl_peer: sip_set_udptl_peer,
+};
+
/*! \brief Convert transfer status to string */
static char *referstatus2str(enum referstatus rstatus)
{
@@ -2387,6 +2453,24 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
r->vrtp = NULL;
}
r->prefs = peer->prefs;
+ if (ast_test_flag(&r->flags[1], SIP_PAGE2_T38SUPPORT)) {
+ r->t38.capability = global_t38_capability;
+ if (r->udptl) {
+ if (ast_udptl_get_error_correction_scheme(r->udptl) == UDPTL_ERROR_CORRECTION_FEC )
+ r->t38.capability |= T38FAX_UDP_EC_FEC;
+ else if (ast_udptl_get_error_correction_scheme(r->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
+ r->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
+ else if (ast_udptl_get_error_correction_scheme(r->udptl) == UDPTL_ERROR_CORRECTION_NONE )
+ r->t38.capability |= T38FAX_UDP_EC_NONE;
+ r->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", r->t38.capability);
+ }
+ r->t38.jointcapability = r->t38.capability;
+ } else if (r->udptl) {
+ ast_udptl_destroy(r->udptl);
+ r->udptl = NULL;
+ }
natflags = ast_test_flag(&r->flags[0], SIP_NAT) & SIP_NAT_ROUTE;
if (r->rtp) {
if (option_debug)
@@ -2400,6 +2484,11 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
ast_rtp_setnat(r->vrtp, natflags);
ast_rtp_setdtmf(r->vrtp, 0);
}
+ if (r->udptl) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", natflags ? "On" : "Off");
+ ast_udptl_setnat(r->udptl, natflags);
+ }
ast_string_field_set(r, peername, peer->username);
ast_string_field_set(r, authname, peer->username);
ast_string_field_set(r, username, peer->username);
@@ -2563,7 +2652,12 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
} else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER_REPLACES")) {
/* We're replacing a call. */
p->options->replaces = ast_var_value(current);
+ } else if (!strcasecmp(ast_var_name(current),"T38CALL")) {
+ p->t38.state = T38_LOCAL_DIRECT;
+ if (option_debug)
+ ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
}
+
}
res = 0;
@@ -2588,6 +2682,9 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
if ( res != -1 ) {
p->callingpres = ast->cid.cid_pres;
p->jointcapability = p->capability;
+ p->t38.jointcapability = p->t38.capability;
+ if (option_debug)
+ ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
if (p->maxtime) {
/* Initialize auto-congest time */
@@ -2655,6 +2752,8 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
ast_rtp_destroy(p->rtp);
if (p->vrtp)
ast_rtp_destroy(p->vrtp);
+ if (p->udptl)
+ ast_udptl_destroy(p->udptl);
if (p->refer)
free(p->refer);
if (p->route) {
@@ -3182,7 +3281,14 @@ static int sip_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
- res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+ if (p->t38.state == T38_PEER_DIRECT) {
+ p->t38.state = T38_ENABLED;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
+ res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+ } else {
+ res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+ }
}
ast_mutex_unlock(&p->lock);
return res;
@@ -3237,6 +3343,21 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
case AST_FRAME_IMAGE:
return 0;
break;
+ case AST_FRAME_MODEM:
+ if (p) {
+ ast_mutex_lock(&p->lock);
+ if (p->udptl) {
+ if ((ast->_state != AST_STATE_UP) &&
+ !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
+ !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_RELIABLE);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ }
+ res = ast_udptl_write(p->udptl, frame);
+ }
+ ast_mutex_unlock(&p->lock);
+ }
+ break;
default:
ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
return 0;
@@ -3504,6 +3625,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
tmp->fds[2] = ast_rtp_fd(i->vrtp);
tmp->fds[3] = ast_rtcp_fd(i->vrtp);
}
+ if (i->udptl) {
+ tmp->fds[5] = ast_udptl_fd(i->udptl);
+ }
if (state == AST_STATE_RING)
tmp->rings = 1;
tmp->adsicpe = AST_ADSI_UNAVAILABLE;
@@ -3697,7 +3821,7 @@ static const char *get_header(const struct sip_request *req, const char *name)
}
/*! \brief Read RTP from network */
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p)
+static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
{
/* Retrieve audio/etc from channel. Assumes p->lock is already held. */
struct ast_frame *f;
@@ -3720,6 +3844,9 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
case 3:
f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */
break;
+ case 5:
+ f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
+ break;
default:
f = &ast_null_frame;
}
@@ -3740,8 +3867,15 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
}
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
f = ast_dsp_process(p->owner, p->vad, f);
- if (option_debug && f && (f->frametype == AST_FRAME_DTMF))
- ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass);
+ if (f && f->frametype == AST_FRAME_DTMF) {
+ if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
+ *faxdetect = 1;
+ } else if (option_debug) {
+ ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass);
+ }
+ }
}
}
}
@@ -3753,10 +3887,31 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
{
struct ast_frame *fr;
struct sip_pvt *p = ast->tech_pvt;
+ int faxdetected = 0;
ast_mutex_lock(&p->lock);
- fr = sip_rtp_read(ast, p);
+ fr = sip_rtp_read(ast, p, &faxdetected);
p->lastrtprx = time(NULL);
+
+ /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
+ /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
+ if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
+ if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+ if (!p->pendinginvite) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
+ p->t38.state = T38_LOCAL_REINVITE;
+ transmit_reinvite_with_t38_sdp(p);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s", p->t38.state, ast->name);
+ }
+ } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
+ }
+
ast_mutex_unlock(&p->lock);
return fr;
}
@@ -3850,6 +4005,8 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
/* If the global videosupport flag is on, we always create a RTP interface for video */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
+ p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) {
ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n",
ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno));
@@ -3867,6 +4024,9 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_rtp_settos(p->vrtp, global_tos_video);
ast_rtp_setdtmf(p->vrtp, 0);
}
+ if (p->udptl) {
+ ast_udptl_settos(p->udptl, global_tos_audio);
+ }
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive;
@@ -3883,6 +4043,8 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_rtp_setnat(p->rtp, natflags);
if (p->vrtp)
ast_rtp_setnat(p->vrtp, natflags);
+ if (p->udptl)
+ ast_udptl_setnat(p->udptl, natflags);
}
if (p->method != SIP_REGISTER)
@@ -3899,6 +4061,17 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
+ if (p->udptl) {
+ p->t38.capability = global_t38_capability;
+ if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY)
+ p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
+ else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC)
+ p->t38.capability |= T38FAX_UDP_EC_FEC;
+ else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE)
+ p->t38.capability |= T38FAX_UDP_EC_NONE;
+ p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+ p->t38.jointcapability = p->t38.capability;
+ }
ast_string_field_set(p, context, default_context);
/* Add to active dialog list */
@@ -4256,6 +4429,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int len = -1;
int portno = -1; /*!< RTP Audio port number */
int vportno = -1; /*!< RTP Video port number */
+ int udptlportno = -1;
+ int peert38capability = 0;
+ char s[256];
+ int old = 0;
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
int peercapability, peernoncodeccapability;
@@ -4366,6 +4543,20 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_set_m_type(newvideortp, codec);
}
+ } else if (p->udptl && (sscanf(m, "image %d udptl t38 %n", &x, &len) == 1)) {
+ if (debug)
+ ast_verbose("Got T.38 offer in SDP\n");
+ udptlportno = x;
+
+ if (p->owner && p->lastinvite) {
+ p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" );
+ } else {
+ p->t38.state = T38_PEER_DIRECT; /* T38 Offered directly from peer in first invite */
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ }
} else
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
if (numberofports > 1)
@@ -4388,7 +4579,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
}
- if (portno == -1 && vportno == -1)
+ if (portno == -1 && vportno == -1 && udptlportno == -1)
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
return -2;
@@ -4404,13 +4595,36 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if (vhp)
memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
-
- /* Setup audio port number */
- sin.sin_port = htons(portno);
+ if (p->rtp) {
+ if (portno > 0) {
+ sin.sin_port = htons(portno);
+ ast_rtp_set_peer(p->rtp, &sin);
+ if (debug)
+ ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ } else {
+ ast_rtp_stop(p->rtp);
+ if (debug)
+ ast_verbose("Peer doesn't provide audio\n");
+ }
+ }
/* Setup video port number */
if (vportno != -1)
vsin.sin_port = htons(vportno);
+ /* Setup UDPTL port number */
+ if (p->udptl) {
+ if (udptlportno > 0) {
+ sin.sin_port = htons(udptlportno);
+ 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(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ } else {
+ ast_udptl_stop(p->udptl);
+ if (debug)
+ ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n");
+ }
+ }
+
/* Next, scan through each "a=rtpmap:" line, noting each
* specified RTP payload type (with corresponding MIME subtype):
*/
@@ -4474,6 +4688,123 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if (p->vrtp)
ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype);
}
+
+ if (udptlportno != -1) {
+ int found = 0, x;
+
+ old = 0;
+
+ /* Scan trough the a= lines for T38 attributes and set apropriate fileds */
+ iterator = req->sdp_start;
+ while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
+ if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"MaxBufferSize:%d\n",x);
+ }
+ if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x);
+ switch (x) {
+ case 14400:
+ peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 12000:
+ peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 9600:
+ peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 7200:
+ peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 4800:
+ peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 2400:
+ peert38capability |= T38FAX_RATE_2400;
+ break;
+ }
+ }
+ if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"FaxVersion: %d\n",x);
+ if (x == 0)
+ peert38capability |= T38FAX_VERSION_0;
+ else if (x == 1)
+ peert38capability |= T38FAX_VERSION_1;
+ }
+ if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"FaxMaxDatagram: %d\n",x);
+ ast_udptl_set_far_max_datagram(p->udptl, x);
+ ast_udptl_set_local_max_datagram(p->udptl, x);
+ }
+ if ((sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"FillBitRemoval: %d\n",x);
+ if (x == 1)
+ peert38capability |= T38FAX_FILL_BIT_REMOVAL;
+ }
+ if ((sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"Transcoding MMR: %d\n",x);
+ if (x == 1)
+ peert38capability |= T38FAX_TRANSCODING_MMR;
+ }
+ if ((sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"Transcoding JBIG: %d\n",x);
+ if (x == 1)
+ peert38capability |= T38FAX_TRANSCODING_JBIG;
+ }
+ if ((sscanf(a, "T38FaxRateManagement:%s", s) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"RateMangement: %s\n", s);
+ if (!strcasecmp(s, "localTCF"))
+ peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF;
+ else if (!strcasecmp(s, "transferredTCF"))
+ peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+ }
+ if ((sscanf(a, "T38FaxUdpEC:%s", s) == 1)) {
+ found = 1;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"UDP EC: %s\n", s);
+ if (!strcasecmp(s, "t38UDPRedundancy")) {
+ peert38capability |= T38FAX_UDP_EC_REDUNDANCY;
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
+ } else if (!strcasecmp(s, "t38UDPFEC")) {
+ peert38capability |= T38FAX_UDP_EC_FEC;
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
+ } else {
+ peert38capability |= T38FAX_UDP_EC_NONE;
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
+ }
+ }
+ }
+ if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */
+ p->t38.peercapability = peert38capability;
+ p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */
+ peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400);
+ p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */
+ }
+ if (debug)
+ ast_log(LOG_DEBUG,"Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n",
+ p->t38.capability,
+ p->t38.peercapability,
+ p->t38.jointcapability);
+ } else {
+ p->t38.state = T38_DISABLED;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ }
/* Now gather all of the codecs that we are asked for: */
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
@@ -5209,6 +5540,142 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
}
+/*! \brief Get Max T.38 Transmision rate from T38 capabilities */
+static int t38_get_rate(int t38cap)
+{
+ int maxrate = (t38cap & (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400));
+
+ if (maxrate & T38FAX_RATE_14400) {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38MaxFaxRate 14400 found\n");
+ return 14400;
+ } else if (maxrate & T38FAX_RATE_12000) {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38MaxFaxRate 12000 found\n");
+ return 12000;
+ } else if (maxrate & T38FAX_RATE_9600) {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38MaxFaxRate 9600 found\n");
+ return 9600;
+ } else if (maxrate & T38FAX_RATE_7200) {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38MaxFaxRate 7200 found\n");
+ return 7200;
+ } else if (maxrate & T38FAX_RATE_4800) {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38MaxFaxRate 4800 found\n");
+ return 4800;
+ } else if (maxrate & T38FAX_RATE_2400) {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38MaxFaxRate 2400 found\n");
+ return 2400;
+ } else {
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "Strange, T38MaxFaxRate NOT found in peers T38 SDP.\n");
+ return 0;
+ }
+}
+
+/*! \brief Add T.38 Session Description Protocol message */
+static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p)
+{
+ int len = 0;
+ int x = 0;
+ struct sockaddr_in udptlsin;
+ char v[256] = "";
+ char s[256] = "";
+ char o[256] = "";
+ char c[256] = "";
+ char t[256] = "";
+ char m_modem[256];
+ char a_modem[1024];
+ char *m_modem_next = m_modem;
+ size_t m_modem_left = sizeof(m_modem);
+ char *a_modem_next = a_modem;
+ size_t a_modem_left = sizeof(a_modem);
+ char iabuf[INET_ADDRSTRLEN];
+ struct sockaddr_in udptldest = { 0, };
+ int debug;
+
+ debug = sip_debug_test_pvt(p);
+ len = 0;
+ if (!p->udptl) {
+ ast_log(LOG_WARNING, "No way to add SDP without an UDPTL structure\n");
+ return -1;
+ }
+
+ if (!p->sessionid) {
+ p->sessionid = getpid();
+ p->sessionversion = p->sessionid;
+ } else
+ p->sessionversion++;
+
+ /* Our T.38 end is */
+ ast_udptl_get_us(p->udptl, &udptlsin);
+
+ /* Determine T.38 UDPTL destination */
+ if (p->udptlredirip.sin_addr.s_addr) {
+ udptldest.sin_port = p->udptlredirip.sin_port;
+ udptldest.sin_addr = p->udptlredirip.sin_addr;
+ } else {
+ udptldest.sin_addr = p->ourip;
+ udptldest.sin_port = udptlsin.sin_port;
+ }
+
+ if (debug) {
+ ast_log(LOG_DEBUG, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(udptlsin.sin_port));
+ }
+
+ /* We break with the "recommendation" and send our IP, in order that our
+ peer doesn't have to ast_gethostbyname() us */
+
+ if (debug) {
+ ast_log(LOG_DEBUG, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n",
+ p->t38.capability,
+ p->t38.peercapability,
+ p->t38.jointcapability);
+ }
+ snprintf(v, sizeof(v), "v=0\r\n");
+ snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), udptldest.sin_addr));
+ snprintf(s, sizeof(s), "s=session\r\n");
+ snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), udptldest.sin_addr));
+ snprintf(t, sizeof(t), "t=0 0\r\n");
+ ast_build_string(&m_modem_next, &m_modem_left, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port));
+
+ if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:0\r\n");
+ if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1)
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxVersion:1\r\n");
+ if ((x = t38_get_rate(p->t38.jointcapability)))
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38MaxBitRate:%d\r\n",x);
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxFillBitRemoval:%d\r\n", (p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) ? 1 : 0);
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingMMR:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_MMR) ? 1 : 0);
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingJBIG:%d\r\n", (p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) ? 1 : 0);
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF");
+ x = ast_udptl_get_local_max_datagram(p->udptl);
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxBuffer:%d\r\n",x);
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxMaxDatagram:%d\r\n",x);
+ if (p->t38.jointcapability != T38FAX_UDP_EC_NONE)
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC");
+ if (p->udptl)
+ len = strlen(m_modem) + strlen(a_modem);
+ add_header(resp, "Content-Type", "application/sdp");
+ add_header_contentLength(resp, len);
+ add_line(resp, v);
+ add_line(resp, o);
+ add_line(resp, s);
+ add_line(resp, c);
+ add_line(resp, t);
+ add_line(resp, m_modem);
+ add_line(resp, a_modem);
+
+ /* Update lastrtprx when we send our SDP */
+ p->lastrtprx = p->lastrtptx = time(NULL);
+
+ return 0;
+}
+
+
/*! \brief Add RFC 2833 DTMF offer to SDP */
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
@@ -5306,6 +5773,11 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
}
+
+ if ((ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP))) {
+ ast_build_string(&m_audio_next, &m_audio_left, " %d", 191);
+ ast_build_string(&a_audio_next, &a_audio_left, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
+ }
/* Check if we need video in this call */
if((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
@@ -5494,6 +5966,26 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
return 0;
}
+/*--- transmit_response_with_t38_sdp: Used for 200 OK and 183 early media ---*/
+static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
+{
+ struct sip_request resp;
+ int seqno;
+
+ if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
+ ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
+ return -1;
+ }
+ respprep(&resp, p, msg, req);
+ if (p->udptl) {
+ ast_udptl_offered_from_local(p->udptl, 0);
+ add_t38_sdp(&resp, p);
+ } else {
+ ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
+ }
+ return send_response(p, &resp, retrans, seqno);
+}
+
/*! \brief copy SIP request (mostly used to save request for responses) */
static void copy_request(struct sip_request *dst, const struct sip_request *src)
{
@@ -5595,6 +6087,31 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
return send_request(p, &req, 1, p->ocseq);
}
+/*--- transmit_reinvite_with_t38_sdp: Transmit reinvite with T38 SDP ---*/
+/* A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
+ INVITE that opened the SIP dialogue
+ We reinvite so that the T38 processing can take place.
+ SIP Signalling stays with * in the path.
+*/
+static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p)
+{
+ struct sip_request req;
+
+ reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
+
+ add_header(&req, "Allow", ALLOWED_METHODS);
+ add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
+ if (sipdebug)
+ add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)");
+ ast_udptl_offered_from_local(p->udptl, 1);
+ add_t38_sdp(&req, p);
+ /* Use this as the basis */
+ initialize_initreq(p, &req);
+ p->lastinvite = p->ocseq;
+ ast_set_flag(&p->flags[0], SIP_OUTGOING);
+ return send_request(p, &req, 1, p->ocseq);
+}
+
/*! \brief Check Contact: URI of SIP message */
static void extract_uri(struct sip_pvt *p, struct sip_request *req)
{
@@ -5924,8 +6441,15 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
}
}
}
- if (sdp && p->rtp) {
- add_sdp(&req, p);
+ if (sdp) {
+ if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
+ ast_udptl_offered_from_local(p->udptl, 1);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ add_t38_sdp(&req, p);
+ } else if (p->rtp) {
+ add_sdp(&req, p);
+ }
} else {
add_header_contentLength(&req, 0);
}
@@ -8077,6 +8601,11 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", usenatroute ? "On" : "Off");
ast_rtp_setnat(p->vrtp, usenatroute);
}
+ if (p->udptl) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", usenatroute ? "On" : "Off");
+ ast_udptl_setnat(p->udptl, usenatroute);
+ }
if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) {
sip_cancel_destroy(p);
ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
@@ -8123,7 +8652,8 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
p->noncodeccapability |= AST_RTP_DTMF;
else
p->noncodeccapability &= ~AST_RTP_DTMF;
-
+ if (p->t38.peercapability)
+ p->t38.jointcapability &= p->t38.peercapability;
p->maxcallbitrate = user->maxcallbitrate;
/* If we do not support video, remove video from call structure */
if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
@@ -8184,6 +8714,10 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", usenatroute ? "On" : "Off");
ast_rtp_setnat(p->vrtp, usenatroute);
}
+ if (p->udptl) {
+ ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", usenatroute ? "On" : "Off");
+ ast_udptl_setnat(p->udptl, usenatroute);
+ }
ast_string_field_set(p, peersecret, peer->secret);
ast_string_field_set(p, peermd5secret, peer->md5secret);
ast_string_field_set(p, subscribecontext, peer->subscribecontext);
@@ -8255,6 +8789,8 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
p->noncodeccapability |= AST_RTP_DTMF;
else
p->noncodeccapability &= ~AST_RTP_DTMF;
+ if (p->t38.peercapability)
+ p->t38.jointcapability &= p->t38.peercapability;
}
ASTOBJ_UNREF(peer, sip_destroy_peer);
} else {
@@ -9055,6 +9591,9 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message
ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT), ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)));
ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
ast_cli(fd, " ACL : %s\n", (peer->ha?"Yes":"No"));
+ ast_cli(fd, " T38 pt UDPTL : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)?"Yes":"No");
+ ast_cli(fd, " T38 pt RTP : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP)?"Yes":"No");
+ ast_cli(fd, " T38 pt TCP : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP)?"Yes":"No");
ast_cli(fd, " CanReinvite : %s\n", ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Yes":"No");
ast_cli(fd, " PromiscRedir : %s\n", ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Yes":"No");
ast_cli(fd, " User=Phone : %s\n", ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Yes":"No");
@@ -9324,6 +9863,9 @@ static int sip_show_settings(int fd, int argc, char *argv[])
ast_cli(fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
ast_cli(fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
ast_cli(fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
+ ast_cli(fd, " T38 fax pt UDPTL: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL) ? "Yes" : "No");
+ ast_cli(fd, " T38 fax pt RTP: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP) ? "Yes" : "No");
+ ast_cli(fd, " T38 fax pt TCP: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP) ? "Yes" : "No");
if (!realtimepeers && !realtimeusers)
ast_cli(fd, " SIP realtime: Disabled\n" );
else
@@ -10476,6 +11018,11 @@ static int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *
ast_copy_string(buf, p->useragent, len);
} else if (!strcasecmp(data, "peername")) {
ast_copy_string(buf, p->peername, len);
+ } else if (!strcasecmp(data, "t38passthrough")) {
+ if (p->t38.state == T38_DISABLED)
+ ast_copy_string(buf, "0", sizeof("0"));
+ else /* T38 is offered or enabled in this call */
+ ast_copy_string(buf, "1", sizeof("1"));
} else {
ast_channel_unlock(chan);
return -1;
@@ -10498,6 +11045,7 @@ static struct ast_custom_function sipchaninfo_function = {
"- uri The URI from the Contact: header.\n"
"- useragent The useragent.\n"
"- peername The name of the peer.\n"
+ "- t38passthrough 1 if T38 is offered or enabled in this channel, otherwise 0\n"
};
/*! \brief Parse 302 Moved temporalily response */
@@ -10672,6 +11220,51 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
build_route(p, req, 1);
}
+ if (p->owner && (p->owner->_state == AST_STATE_UP)) { /* if this is a re-invite */
+ struct ast_channel *bridgepeer = NULL;
+ struct sip_pvt *bridgepvt = NULL;
+
+ bridgepeer = ast_bridged_channel(p->owner);
+ if (!bridgepeer->tech) {
+ ast_log(LOG_WARNING, "Ooooh.. no tech! That's REALLY bad\n");
+ break;
+ }
+ if (!strcasecmp(bridgepeer->tech->type,"SIP")) {
+ bridgepvt = (struct sip_pvt*)(bridgepeer->tech_pvt);
+ if (bridgepvt->udptl) {
+ if (p->t38.state == T38_PEER_REINVITE) {
+ sip_handle_t38_reinvite(bridgepeer, p, 0);
+ } else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
+ ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
+ /* Insted of this we should somehow re-invite the other side of the bridge to RTP */
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ }
+ } else {
+ ast_log(LOG_WARNING, "Strange... The other side of the bridge don't have udptl struct\n");
+ ast_mutex_lock(&bridgepvt->lock);
+ bridgepvt->t38.state = T38_DISABLED;
+ ast_mutex_unlock(&bridgepvt->lock);
+ if (option_debug)
+ ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->tech->type);
+ p->t38.state = T38_DISABLED;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ }
+ } else {
+ /* Other side is not a SIP channel */
+ ast_log(LOG_WARNING, "Strange... The other side of the bridge is not a SIP channel\n");
+ p->t38.state = T38_DISABLED;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ }
+ }
+ if ((p->t38.state == T38_LOCAL_REINVITE) || (p->t38.state == T38_LOCAL_DIRECT)) {
+ /* If there was T38 reinvite and we are supposed to answer with 200 OK than this should set us to T38 negotiated mode */
+ p->t38.state = T38_ENABLED;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ }
+
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
@@ -11164,6 +11757,10 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
/* Immediately stop VRTP */
ast_rtp_stop(p->vrtp);
}
+ if (p->udptl) {
+ /* Immediately stop UDPTL */
+ ast_udptl_stop(p->udptl);
+ }
/* XXX Locking issues?? XXX */
switch(resp) {
case 300: /* Multiple Choices */
@@ -12280,8 +12877,74 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
case AST_STATE_UP:
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s: New call is UP.... \n", c->name);
- transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
-
+ if (p->t38.state == T38_PEER_REINVITE) {
+ struct ast_channel *bridgepeer = NULL;
+ struct sip_pvt *bridgepvt = NULL;
+
+ if ((bridgepeer = ast_bridged_channel(p->owner))) {
+ /* We have a bridge, and this is re-invite to switchover to T38 so we send re-invite with T38 SDP, to other side of bridge*/
+ /*! XXX: we should also check here does the other side supports t38 at all !!! XXX */
+ if (!strcasecmp(bridgepeer->tech->type, "SIP")) { /* If we are bridged to SIP channel */
+ bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
+ if (bridgepvt->t38.state == T38_DISABLED) {
+ if (bridgepvt->udptl) { /* If everything is OK with other side's udptl struct */
+ /* Send re-invite to the bridged channel */
+ sip_handle_t38_reinvite(bridgepeer, p, 1);
+ } else { /* Something is wrong with peers udptl struct */
+ ast_log(LOG_WARNING, "Strange... The other side of the bridge don't have udptl struct\n");
+ ast_mutex_lock(&bridgepvt->lock);
+ bridgepvt->t38.state = T38_DISABLED;
+ ast_mutex_unlock(&bridgepvt->lock);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->name);
+ if (ast_test_flag(req, SIP_PKT_IGNORE))
+ transmit_response(p, "488 Not acceptable here", req);
+ else
+ transmit_response_reliable(p, "488 Not acceptable here", req);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ }
+ }
+ } else {
+ /* Other side is not a SIP channel */
+ if (ast_test_flag(req, SIP_PKT_IGNORE))
+ transmit_response(p, "488 Not acceptable here", req);
+ else
+ transmit_response_reliable(p, "488 Not acceptable here", req);
+ p->t38.state = T38_DISABLED;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ }
+ } else {
+ /* we are not bridged in a call */
+ transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL);
+ p->t38.state = T38_ENABLED;
+ if (option_debug)
+ ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
+ }
+ } else if (p->t38.state == T38_DISABLED) { /* Channel doesn't have T38 offered or enabled */
+ /* If we are bridged to a channel that has T38 enabled than this is a case of RTP re-invite after T38 session */
+ /* so handle it here (re-invite other party to RTP) */
+ struct ast_channel *bridgepeer = NULL;
+ struct sip_pvt *bridgepvt = NULL;
+ if ((bridgepeer = ast_bridged_channel(p->owner))) {
+ if (!strcasecmp(bridgepeer->tech->type, sip_tech.type)) {
+ bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
+ if (bridgepvt->t38.state == T38_ENABLED) {
+ ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
+ /* Insted of this we should somehow re-invite the other side of the bridge to RTP */
+ if (ast_test_flag(req, SIP_PKT_IGNORE))
+ transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
+ else
+ transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ } else {
+ /* No bridged peer with T38 enabled*/
+ transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
+ }
+ }
+ }
+ }
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@@ -12780,6 +13443,10 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
/* Immediately stop VRTP */
ast_rtp_stop(p->vrtp);
}
+ if (p->udptl) {
+ /* Immediately stop UDPTL */
+ ast_udptl_stop(p->udptl);
+ }
if (p->owner)
ast_queue_hangup(p->owner);
else
@@ -12835,6 +13502,10 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
/* Immediately stop VRTP */
ast_rtp_stop(p->vrtp);
}
+ if (p->udptl) {
+ /* Immediately stop UDPTL */
+ ast_udptl_stop(p->udptl);
+ }
if (!ast_strlen_zero(get_header(req, "Also"))) {
ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr));
@@ -14063,6 +14734,15 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} else if (!strcasecmp(v->name, "allowsubscribe")) {
ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
+ } else if (!strcasecmp(v->name, "t38pt_udptl")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_UDPTL);
+ } else if (!strcasecmp(v->name, "t38pt_rtp")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_RTP);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_RTP);
+ } else if (!strcasecmp(v->name, "t38pt_tcp")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_TCP);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_TCP);
}
return res;
@@ -14309,7 +14989,22 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
user->maxcallbitrate = atoi(v->value);
if (user->maxcallbitrate < 0)
user->maxcallbitrate = default_maxcallbitrate;
- }
+ } else if (!strcasecmp(v->name, "t38pt_udptl")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+ } else
+ ast_clear_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+ } else if (!strcasecmp(v->name, "t38pt_rtp")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+ } else
+ ast_clear_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+ } else if (!strcasecmp(v->name, "t38pt_tcp")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+ } else
+ ast_clear_flag(&user->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+ }
}
ast_copy_flags(&user->flags[0], &userflags[0], mask[0].flags);
ast_copy_flags(&user->flags[1], &userflags[1], mask[1].flags);
@@ -14591,6 +15286,21 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
peer->maxcallbitrate = atoi(v->value);
if (peer->maxcallbitrate < 0)
peer->maxcallbitrate = default_maxcallbitrate;
+ } else if (!strcasecmp(v->name, "t38pt_udptl")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+ } else
+ ast_clear_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+ } else if (!strcasecmp(v->name, "t38pt_rtp")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+ } else
+ ast_clear_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+ } else if (!strcasecmp(v->name, "t38pt_tcp")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+ } else
+ ast_clear_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP);
}
}
if (!ast_test_flag(&global_flags[1], SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC) && realtime) {
@@ -14952,6 +15662,18 @@ static int reload_config(enum channelreloadreason reason)
default_maxcallbitrate = atoi(v->value);
if (default_maxcallbitrate < 0)
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
+ } else if (!strcasecmp(v->name, "t38pt_udptl")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
+ }
+ } else if (!strcasecmp(v->name, "t38pt_rtp")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP);
+ }
+ } else if (!strcasecmp(v->name, "t38pt_tcp")) {
+ if (ast_true(v->value)) {
+ ast_set_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP);
+ }
}
}
@@ -15093,6 +15815,131 @@ static int reload_config(enum channelreloadreason reason)
return 0;
}
+static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
+{
+ struct sip_pvt *p;
+ struct ast_udptl *udptl = NULL;
+
+ p = chan->tech_pvt;
+ if (!p)
+ return NULL;
+
+ ast_mutex_lock(&p->lock);
+ if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ udptl = p->udptl;
+ ast_mutex_unlock(&p->lock);
+ return udptl;
+}
+
+static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
+{
+ struct sip_pvt *p;
+
+ p = chan->tech_pvt;
+ if (!p)
+ return -1;
+ ast_mutex_lock(&p->lock);
+ if (udptl)
+ ast_udptl_get_peer(udptl, &p->udptlredirip);
+ else
+ memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+ if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+ if (!p->pendinginvite) {
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
+ }
+ transmit_reinvite_with_t38_sdp(p);
+ } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf),udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
+ }
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
+ }
+ /* Reset lastrtprx timer */
+ p->lastrtprx = p->lastrtptx = time(NULL);
+ ast_mutex_unlock(&p->lock);
+ return 0;
+}
+
+static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite)
+{
+ struct sip_pvt *p;
+ int flag = 0;
+
+ p = chan->tech_pvt;
+ if (!p || !pvt->udptl)
+ return -1;
+
+ /* Setup everything on the other side like offered/responded from first side */
+ ast_mutex_lock(&p->lock);
+ p->t38.jointcapability = p->t38.peercapability = pvt->t38.jointcapability;
+ ast_udptl_set_far_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
+ ast_udptl_set_local_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
+ ast_udptl_set_error_correction_scheme(p->udptl, ast_udptl_get_error_correction_scheme(pvt->udptl));
+
+ if (reinvite) { /* If we are handling sending re-invite to the other side of the bridge */
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
+ ast_udptl_get_peer(pvt->udptl, &p->udptlredirip);
+ flag =1;
+ } else {
+ memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+ }
+ if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+ if (!p->pendinginvite) {
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ if (flag)
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+ else
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
+ }
+ transmit_reinvite_with_t38_sdp(p);
+ } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ if (flag)
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+ else
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
+ }
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
+ }
+ /* Reset lastrtprx timer */
+ p->lastrtprx = p->lastrtptx = time(NULL);
+ ast_mutex_unlock(&p->lock);
+ return 0;
+ } else { /* If we are handling sending 200 OK to the other side of the bridge */
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
+ ast_udptl_get_peer(pvt->udptl, &p->udptlredirip);
+ flag = 1;
+ } else {
+ memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+ }
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ if (flag)
+ ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+ else
+ ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
+ }
+ pvt->t38.state = T38_ENABLED;
+ p->t38.state = T38_ENABLED;
+ if (option_debug > 1) {
+ ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>");
+ ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");
+ }
+ transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+ p->lastrtprx = p->lastrtptx = time(NULL);
+ ast_mutex_unlock(&p->lock);
+ return 0;
+ }
+}
+
+
/*! \brief Returns null if we can't reinvite audio (part of RTP interface) */
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
{
@@ -15522,6 +16369,9 @@ static int load_module(void *mod)
/* Tell the RTP subdriver that we're here */
ast_rtp_proto_register(&sip_rtp);
+ /* Tell the UDPTL subdriver that we're here */
+ ast_udptl_proto_register(&sip_udptl);
+
/* Register dialplan applications */
ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader);
@@ -15566,6 +16416,8 @@ static int unload_module(void *mod)
ast_rtp_proto_unregister(&sip_rtp);
+ ast_udptl_proto_unregister(&sip_udptl);
+
ast_manager_unregister("SIPpeers");
ast_manager_unregister("SIPshowpeer");
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-27 17:48:51 +0000