[asterisk-commits] file: branch 1.6.2 r184951 - in /branches/1.6.2: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Mar 30 09:43:22 CDT 2009
Author: file
Date: Mon Mar 30 09:43:12 2009
New Revision: 184951
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=184951
Log:
Merged revisions 184948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines
Merged revisions 184947 via svnmerge from
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/
........
................
Modified:
branches/1.6.2/ (props changed)
branches/1.6.2/channels/chan_sip.c
Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=184951&r1=184950&r2=184951
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Mon Mar 30 09:43:12 2009
@@ -1455,7 +1455,6 @@
/*! \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 */
@@ -1469,6 +1468,7 @@
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 */
@@ -2293,7 +2293,7 @@
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);
@@ -3896,7 +3896,6 @@
/* 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:
@@ -4618,6 +4617,10 @@
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>");
@@ -4997,9 +5000,6 @@
} 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);
}
}
@@ -5769,15 +5769,9 @@
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);
- }
+ 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;
@@ -5814,9 +5808,13 @@
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);
}
- p->lastrtptx = time(NULL);
- res = ast_rtp_write(p->rtp, frame);
}
sip_pvt_unlock(p);
}
@@ -5871,8 +5869,16 @@
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)
- res = ast_udptl_write(p->udptl, frame);
+ 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;
@@ -6293,10 +6299,6 @@
pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
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) {
@@ -8009,7 +8011,7 @@
/* 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) {
@@ -9122,92 +9124,6 @@
}
}
-/*! \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,
@@ -9270,7 +9186,7 @@
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;
@@ -9279,8 +9195,10 @@
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 */
@@ -9289,16 +9207,18 @@
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;
@@ -9329,177 +9249,226 @@
p->sessionversion++;
}
- capability = p->jointcapability;
-
- /* XXX note, Video and Text are negated - 'true' means 'no' */
- ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
- p->novideo ? "True" : "False", p->notext ? "True" : "False");
- ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
+ 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' */
+ ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
+ p->novideo ? "True" : "False", p->notext ? "True" : "False");
+ ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
- if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) {
- ast_str_append(&m_audio, 0, " %d", 191);
- ast_str_append(&a_audio, 0, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
- }
+ if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) {
+ ast_str_append(&m_audio, 0, " %d", 191);
+ ast_str_append(&a_audio, 0, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
+ }
#endif
- /* Check if we need audio */
- if (capability & AST_FORMAT_AUDIO_MASK)
- needaudio = TRUE;
-
- /* Check if we need video in this call */
- if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
- if (p->vrtp) {
- needvideo = TRUE;
- ast_debug(2, "This call needs video offers!\n");
- } else
- 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);
+ /* Check if we need audio */
+ if (capability & AST_FORMAT_AUDIO_MASK)
+ needaudio = TRUE;
+
+ /* Check if we need video in this call */
+ if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
+ if (p->vrtp) {
+ needvideo = TRUE;
+ ast_debug(2, "This call needs video offers!\n");
+ } else
+ ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
+ }
+
+ if (debug)
+ ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(sin.sin_port));
+
+ /* Ok, we need video. Let's add what we need for video and set codecs.
+ Video is handled differently than audio since we can not transcode. */
+ if (needvideo) {
+ ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+
+ /* Build max bitrate string */
+ if (p->maxcallbitrate)
+ snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
+ if (debug)
+ ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(vsin.sin_port));
+ }
+
+ /* Check if we need text in this call */
+ if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
+ if (sipdebug_text)
+ ast_verbose("We think we can do text\n");
+ if (p->trtp) {
+ if (sipdebug_text)
+ ast_verbose("And we have a text rtp object\n");
+ needtext = TRUE;
+ ast_debug(2, "This call needs text offers! \n");
+ } else
+ ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
+ }
- if (debug)
- ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(sin.sin_port));
-
- /* Ok, we need video. Let's add what we need for video and set codecs.
- Video is handled differently than audio since we can not transcode. */
- if (needvideo) {
- ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
-
- /* Build max bitrate string */
- if (p->maxcallbitrate)
- snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
- if (debug)
- ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(vsin.sin_port));
- }
-
- /* Check if we need text in this call */
- if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
- if (sipdebug_text)
- ast_verbose("We think we can do text\n");
- if (p->trtp) {
+ /* Ok, we need text. Let's add what we need for text and set codecs.
+ Text is handled differently than audio since we can not transcode. */
+ if (needtext) {
if (sipdebug_text)
- ast_verbose("And we have a text rtp object\n");
- needtext = TRUE;
- ast_debug(2, "This call needs text offers! \n");
- } else
- ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
- }
-
- /* Ok, we need text. Let's add what we need for text and set codecs.
- Text is handled differently than audio since we can not transcode. */
- if (needtext) {
- if (sipdebug_text)
- ast_verbose("Lets set up the text sdp\n");
- /* Determine text destination */
- if (p->tredirip.sin_addr.s_addr) {
- tdest.sin_addr = p->tredirip.sin_addr;
- tdest.sin_port = p->tredirip.sin_port;
+ ast_verbose("Lets set up the text sdp\n");
+ /* Determine text destination */
+ if (p->tredirip.sin_addr.s_addr) {
+ tdest.sin_addr = p->tredirip.sin_addr;
+ tdest.sin_port = p->tredirip.sin_port;
+ } else {
+ tdest.sin_addr = p->ourip.sin_addr;
+ 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));
+
+ }
+
+ /* Start building generic SDP headers */
+
+ /* We break with the "recommendation" and send our IP, in order that our
+ peer doesn't have to ast_gethostbyname() us */
+
+ 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)
+ hold = "a=recvonly\r\n";
+ else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
+ hold = "a=inactive\r\n";
+ else
+ hold = "a=sendrecv\r\n";
+
+ /* Now, start adding audio codecs. These are added in this order:
+ - First what was requested by the calling channel
+ - Then preferences in order from sip.conf device config for this peer/user
+ - Then other codecs in capabilities, including video
+ */
+
+ /* Prefer the audio codec we were requested to use, first, no matter what
+ Note that p->prefcodec can include video codecs, so mask them out
+ */
+ if (capability & p->prefcodec) {
+ int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
+
+ add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ alreadysent |= codec;
+ }
+
+ /* Start by sending our preferred audio/video codecs */
+ for (x = 0; x < 32; x++) {
+ int codec;
+
+ if (!(codec = ast_codec_pref_index(&p->prefs, x)))
+ break;
+
+ if (!(capability & codec))
+ continue;
+
+ if (alreadysent & codec)
+ continue;
+
+ add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ alreadysent |= codec;
+ }
+
+ /* Now send any other common audio and video codecs, and non-codec formats: */
+ for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
+ if (!(capability & x)) /* Codec not requested */
+ continue;
+
+ if (alreadysent & x) /* Already added to SDP */
+ continue;
+
+ if (x & AST_FORMAT_AUDIO_MASK)
+ add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ else if (x & AST_FORMAT_VIDEO_MASK)
+ add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size);
+ else if (x & AST_FORMAT_TEXT_MASK)
+ add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
+ }
+
+ /* Now add DTMF RFC2833 telephony-event as a codec */
+ for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
+ if (!(p->jointnoncodeccapability & x))
+ continue;
+
+ add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
+ }
+
+ ast_debug(3, "-- Done with adding codecs to SDP\n");
+
+ if (!p->owner || !ast_internal_timing_enabled(p->owner))
+ ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
+
+ if (min_audio_packet_size)
+ ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
+
+ /* XXX don't think you can have ptime for video */
+ if (min_video_packet_size)
+ ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);
+
+ /* XXX don't think you can have ptime for text */
+ if (min_text_packet_size)
+ ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);
+
+ if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 ||
+ 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 {
- tdest.sin_addr = p->ourip.sin_addr;
- 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));
-
- }
-
- /* Start building generic SDP headers */
-
- /* 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)
- hold = "a=recvonly\r\n";
- else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
- hold = "a=inactive\r\n";
- else
- hold = "a=sendrecv\r\n";
-
- /* Now, start adding audio codecs. These are added in this order:
- - First what was requested by the calling channel
- - Then preferences in order from sip.conf device config for this peer/user
- - Then other codecs in capabilities, including video
- */
-
- /* Prefer the audio codec we were requested to use, first, no matter what
- Note that p->prefcodec can include video codecs, so mask them out
- */
- if (capability & p->prefcodec) {
- int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
-
- add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
- alreadysent |= codec;
- }
-
- /* Start by sending our preferred audio/video codecs */
- for (x = 0; x < 32; x++) {
- int codec;
-
- if (!(codec = ast_codec_pref_index(&p->prefs, x)))
- break;
-
- if (!(capability & codec))
- continue;
-
- if (alreadysent & codec)
- continue;
-
- add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
- alreadysent |= codec;
- }
-
- /* Now send any other common audio and video codecs, and non-codec formats: */
- for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
- if (!(capability & x)) /* Codec not requested */
- continue;
-
- if (alreadysent & x) /* Already added to SDP */
- continue;
-
- if (x & AST_FORMAT_AUDIO_MASK)
- add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size);
- else if (x & AST_FORMAT_VIDEO_MASK)
- add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size);
- else if (x & AST_FORMAT_TEXT_MASK)
- add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
- }
-
- /* Now add DTMF RFC2833 telephony-event as a codec */
- for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
- if (!(p->jointnoncodeccapability & x))
- continue;
-
- add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
- }
-
- ast_debug(3, "-- Done with adding codecs to SDP\n");
-
- if (!p->owner || !ast_internal_timing_enabled(p->owner))
- ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
-
- if (min_audio_packet_size)
- ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
-
- /* XXX don't think you can have ptime for video */
- if (min_video_packet_size)
- ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);
-
- /* XXX don't think you can have ptime for text */
- if (min_text_packet_size)
- ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);
-
- if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 ||
- 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");
+ 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");
@@ -9507,6 +9476,8 @@
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);
@@ -9516,6 +9487,8 @@
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);
@@ -9541,6 +9514,10 @@
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 ? */
@@ -9563,7 +9540,7 @@
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)
@@ -9612,8 +9589,12 @@
ast_debug(1, "Setting framing from config on incoming call\n");
ast_rtp_codec_setpref(p->rtp, &p->prefs);
}
- try_suggested_sip_codec(p);
- add_sdp(&resp, p, oldsdp);
+ try_suggested_sip_codec(p);
+ 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)
@@ -9695,9 +9676,9 @@
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);
@@ -10100,12 +10081,12 @@
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);
@@ -16794,20 +16775,6 @@
/* 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)
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