Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

Merged revisions 184947 via svnmerge from

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

........
  r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines
  
  Improve our handling of T38 in the initial INVITE from a device.
  
  We now answer with matching media streams to what is requested. If an INVITE
  is received with both a T38 and RTP media stream this means we answer with both.
  For any outgoing calls created as a result of this inbound one no T38 is requested
  in the initial INVITE. Instead if we start receiving udptl packets we trigger a
  reinvite on the outbound side.
  
  (closes issue #12437)
  Reported by: marsosa
  Tested by: pinga-fogo, okrief, file, afu
  
  Review: http://reviewboard.digium.com/r/208/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/1.8.6
Joshua Colp 17 years ago
parent 3ec3742a97
commit aa056be678
1 changed files with 235 additions and 268 deletions
Show Stats Download Patch File Download Diff File

237
channels/chan_sip.c
Unescape View File

@ -1456,7 +1456,6 @@ static int sipdebug_text;
/*! \brief T38 States 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 */
@ -1470,6 +1469,7 @@ struct t38properties {
int peercapability; /*!< Peers T38 capability */
int jointcapability; /*!< Supported T38 capability at both ends */
enum t38state state; /*!< T.38 state */
unsigned int direct:1; /*!< Whether the T38 came from the initial invite or not */
};
/*! \brief Parameters to know status of transfer */
@ -2294,7 +2294,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38);
static void do_setnat(struct sip_pvt *p, int natflags);
static void stop_media_flows(struct sip_pvt *p);
@ -3897,7 +3897,6 @@ static int sip_queryoption(struct ast_channel *chan, int option, void *data, int
/* Now if T38 support is enabled we need to look and see what the current state is to get what we want to report back */
if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT)) {
switch (p->t38.state) {
case T38_LOCAL_DIRECT:
case T38_LOCAL_REINVITE:
case T38_PEER_DIRECT:
case T38_PEER_REINVITE:
@ -4619,6 +4618,10 @@ static void change_t38_state(struct sip_pvt *p, int state)
if (old == state)
return;
if (state == T38_PEER_DIRECT) {
p->t38.direct = 1;
}
p->t38.state = state;
ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");
@ -4998,9 +5001,6 @@ 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;
ast_debug(1, "T38State change to %d on channel %s\n", p->t38.state, ast->name);
}
}
@ -5770,16 +5770,10 @@ static int sip_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
if (p->t38.state == T38_PEER_DIRECT) {
change_t38_state(p, T38_ENABLED);
res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
} else {
ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
}
sip_pvt_unlock(p);
return res;
}
@ -5815,10 +5809,14 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
} else if (p->t38.state == T38_ENABLED && !p->t38.direct) {
change_t38_state(p, T38_DISABLED);
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else {
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->rtp, frame);
}
}
sip_pvt_unlock(p);
}
break;
@ -5872,8 +5870,16 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
we simply forget the frames if we get modem frames before the bridge is up.
Fax will re-transmit.
*/
if (p->udptl && ast->_state == AST_STATE_UP)
if (ast->_state == AST_STATE_UP) {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && p->t38.state == T38_DISABLED) {
if (!p->pendinginvite) {
change_t38_state(p, T38_LOCAL_REINVITE);
transmit_reinvite_with_sdp(p, TRUE, FALSE);
}
} else if (p->udptl && p->t38.state == T38_ENABLED) {
res = ast_udptl_write(p->udptl, frame);
}
}
sip_pvt_unlock(p);
}
break;
@ -6295,10 +6301,6 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
/* If the INVITE contains T.38 SDP information set the proper channel variable so a created outgoing call will also have T.38 */
if (i->udptl && i->t38.state == T38_PEER_DIRECT)
pbx_builtin_setvar_helper(tmp, "_T38CALL", "1");
/* Set channel variables for this call from configuration */
for (v = i->chanvars ; v ; v = v->next) {
char valuebuf[1024];
@ -8010,7 +8012,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Remote party offers T38, we need to update state */
if (t38action == SDP_T38_ACCEPT) {
if (p->t38.state == T38_LOCAL_DIRECT || p->t38.state == T38_LOCAL_REINVITE)
if (p->t38.state == T38_LOCAL_REINVITE)
change_t38_state(p, T38_ENABLED);
} else if (t38action == SDP_T38_INITIATE) {
if (p->owner && p->lastinvite) {
@ -9127,92 +9129,6 @@ static int t38_get_rate(int t38cap)
}
}
/*! \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;
struct ast_str *m_modem = ast_str_alloca(1024);
struct ast_str *a_modem = ast_str_alloca(1024);
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 = (int)ast_random();
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.sin_addr;
udptldest.sin_port = udptlsin.sin_port;
}
if (debug)
ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), 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_debug(1, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n",
p->t38.capability,
p->t38.peercapability,
p->t38.jointcapability);
}
ast_str_append(&m_modem, 0, "v=0\r\n");
ast_str_append(&m_modem, 0, "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner , p->sessionid, p->sessionversion, ast_inet_ntoa(udptldest.sin_addr));
ast_str_append(&m_modem, 0, "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
ast_str_append(&m_modem, 0, "c=IN IP4 %s\r\n", ast_inet_ntoa(udptldest.sin_addr));
ast_str_append(&m_modem, 0, "t=0 0\r\n");
ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ntohs(udptldest.sin_port));
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
ast_str_append(&a_modem, 0, "a=T38FaxVersion:0\r\n");
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1)
ast_str_append(&a_modem, 0, "a=T38FaxVersion:1\r\n");
if ((x = t38_get_rate(p->t38.jointcapability)))
ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", x);
if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL)
ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR)
ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG)
ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
ast_str_append(&a_modem, 0, "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_str_append(&a_modem, 0, "a=T38FaxMaxBuffer:%d\r\n", x);
ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%d\r\n", x);
if (p->t38.jointcapability != T38FAX_UDP_EC_NONE)
ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC");
len = m_modem->used + a_modem->used;
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
add_line(resp, m_modem->str);
add_line(resp, a_modem->str);
/* 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,
struct ast_str **m_buf, struct ast_str **a_buf,
@ -9275,7 +9191,7 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo,
is used in Session-Timers where RE-INVITEs are used for refreshing SIP sessions
without modifying the media session in any way.
*/
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp)
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
{
int len = 0;
int alreadysent = 0;
@ -9284,8 +9200,10 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
struct sockaddr_in vsin;
struct sockaddr_in tsin;
struct sockaddr_in dest;
struct sockaddr_in udptlsin;
struct sockaddr_in vdest = { 0, };
struct sockaddr_in tdest = { 0, };
struct sockaddr_in udptldest = { 0, };
/* SDP fields */
char *version = "v=0\r\n"; /* Protocol version */
@ -9294,16 +9212,18 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
char connection[256]; /* Connection data */
char *session_time = "t=0 0\r\n"; /* Time the session is active */
char bandwidth[256] = ""; /* Max bitrate */
char *hold;
char *hold = "";
struct ast_str *m_audio = ast_str_alloca(256); /* Media declaration line for audio */
struct ast_str *m_video = ast_str_alloca(256); /* Media declaration line for video */
struct ast_str *m_text = ast_str_alloca(256); /* Media declaration line for text */
struct ast_str *m_modem = ast_str_alloca(256); /* Media declaration line for modem */
struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */
struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */
struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
int x;
int capability;
int capability = 0;
int needaudio = FALSE;
int needvideo = FALSE;
int needtext = FALSE;
@ -9334,6 +9254,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
p->sessionversion++;
}
get_our_media_address(p, needvideo, &sin, &vsin, &tsin, &dest, &vdest);
snprintf(owner, sizeof(owner), "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner, p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
if (add_audio) {
capability = p->jointcapability;
/* XXX note, Video and Text are negated - 'true' means 'no' */
@ -9361,9 +9287,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
}
/* Get our media addresses */
get_our_media_address(p, needvideo, &sin, &vsin, &tsin, &dest, &vdest);
if (debug)
ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(sin.sin_port));
@ -9406,7 +9329,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
tdest.sin_port = tsin.sin_port;
}
ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
if (debug) /* XXX should I use tdest below ? */
ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port));
@ -9417,8 +9339,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
snprintf(owner, sizeof(owner), "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner, p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
@ -9505,6 +9425,55 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 ||
a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
}
if (add_t38) {
/* 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.sin_addr;
udptldest.sin_port = udptlsin.sin_port;
}
if (debug)
ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), 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_debug(1, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n",
p->t38.capability,
p->t38.peercapability,
p->t38.jointcapability);
}
ast_str_append(&m_modem, 0, "m=image %d udptl t38", ntohs(udptldest.sin_port));
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
ast_str_append(&a_modem, 0, "a=T38FaxVersion:0\r\n");
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1)
ast_str_append(&a_modem, 0, "a=T38FaxVersion:1\r\n");
if ((x = t38_get_rate(p->t38.jointcapability)))
ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", x);
if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL)
ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR)
ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG)
ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
ast_str_append(&a_modem, 0, "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_str_append(&a_modem, 0, "a=T38FaxMaxBuffer:%d\r\n", x);
ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%d\r\n", x);
if (p->t38.jointcapability != T38FAX_UDP_EC_NONE)
ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC");
}
if (needaudio)
ast_str_append(&m_audio, 0, "\r\n");
@ -9512,6 +9481,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
ast_str_append(&m_video, 0, "\r\n");
if (needtext)
ast_str_append(&m_text, 0, "\r\n");
if (add_t38)
ast_str_append(&m_modem, 0, "\r\n");
len = strlen(version) + strlen(subject) + strlen(owner) +
strlen(connection) + strlen(session_time);
@ -9521,6 +9492,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
len += m_video->used + a_video->used + strlen(bandwidth) + strlen(hold);
if (needtext) /* only if text response is appropriate */
len += m_text->used + a_text->used + strlen(hold);
if (add_t38)
len += m_modem->used + a_modem->used;
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
@ -9546,6 +9519,10 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
add_line(resp, a_text->str);
add_line(resp, hold); /* Repeat hold for the text stream */
}
if (add_t38) {
add_line(resp, m_modem->str);
add_line(resp, a_modem->str);
}
/* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@ -9568,7 +9545,7 @@ static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct s
respprep(&resp, p, msg, req);
if (p->udptl) {
ast_udptl_offered_from_local(p->udptl, 0);
add_t38_sdp(&resp, p);
add_sdp(&resp, p, 0, 0, 1);
} else
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
if (retrans && !p->pendinginvite)
@ -9618,7 +9595,11 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
ast_rtp_codec_setpref(p->rtp, &p->prefs);
}
try_suggested_sip_codec(p);
add_sdp(&resp, p, oldsdp);
if (p->t38.state == T38_PEER_DIRECT || p->t38.state == T38_ENABLED) {
add_sdp(&resp, p, oldsdp, TRUE, TRUE);
} else {
add_sdp(&resp, p, oldsdp, TRUE, FALSE);
}
} else
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
if (reliable && !p->pendinginvite)
@ -9700,9 +9681,9 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
if (p->do_history)
append_history(p, "ReInv", "Re-invite sent");
if (t38version)
add_t38_sdp(&req, p);
add_sdp(&req, p, oldsdp, FALSE, TRUE);
else
add_sdp(&req, p, oldsdp);
add_sdp(&req, p, oldsdp, TRUE, FALSE);
/* Use this as the basis */
initialize_initreq(p, &req);
@ -10105,12 +10086,12 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
ast_channel_unlock(chan);
}
if (sdp) {
if (p->udptl && (p->t38.state == T38_LOCAL_DIRECT || p->t38.state == T38_LOCAL_REINVITE)) {
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
ast_udptl_offered_from_local(p->udptl, 1);
ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
add_t38_sdp(&req, p);
add_sdp(&req, p, FALSE, FALSE, TRUE);
} else if (p->rtp)
add_sdp(&req, p, FALSE);
add_sdp(&req, p, FALSE, TRUE, FALSE);
} else {
if (!p->notify_headers) {
add_header_contentLength(&req, 0);
@ -16799,20 +16780,6 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
/* Trigger a reinvite back to audio */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
/* We tried to send T.38 out in an initial INVITE and the remote side rejected it,
right now we can't fall back to audio so totally abort.
*/
/* Try to reset RTP timers */
ast_rtp_set_rtptimers_onhold(p->rtp);
ast_log(LOG_ERROR, "Got error on T.38 initial invite. Bailing out.\n");
change_t38_state(p, T38_DISABLED);
/* The dialog is now terminated */
if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
pvt_set_needdestroy(p, "got error on T.38 initial invite");
sip_alreadygone(p);
} else {
/* We can't set up this call, so give up */
if (p->owner && !req->ignore)

Write Preview
Loading…
Cancel
Save