[svn-commits] file: branch 1.4 r184947 - /branches/1.4/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Mar 30 09:36:08 CDT 2009
Author: file
Date: Mon Mar 30 09:35:47 2009
New Revision: 184947
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=184947
Log:
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/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=184947&r1=184946&r2=184947
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Mon Mar 30 09:35:47 2009
@@ -847,7 +847,6 @@
/*! \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 */
@@ -861,6 +860,7 @@
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 */
@@ -1325,7 +1325,7 @@
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,
int debug);
-static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p);
+static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int add_audio, int add_t38);
static void stop_media_flows(struct sip_pvt *p);
/*--- Authentication stuff */
@@ -3053,12 +3053,7 @@
} 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;
@@ -3756,16 +3751,9 @@
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
- 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);
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- } else {
- res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- }
+
+ res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
+ ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
ast_mutex_unlock(&p->lock);
return res;
@@ -3802,9 +3790,13 @@
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ } else if (p->t38.state == T38_ENABLED && !p->t38.direct) {
+ p->t38.state = T38_DISABLED;
+ transmit_reinvite_with_sdp(p);
+ } else {
+ p->lastrtptx = time(NULL);
+ res = ast_rtp_write(p->rtp, frame);
}
- p->lastrtptx = time(NULL);
- res = ast_rtp_write(p->rtp, frame);
}
ast_mutex_unlock(&p->lock);
}
@@ -3837,8 +3829,16 @@
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) {
+ p->t38.state = T38_LOCAL_REINVITE;
+ transmit_reinvite_with_t38_sdp(p);
+ }
+ } else if (p->t38.state == T38_ENABLED) {
+ res = ast_udptl_write(p->udptl, frame);
+ }
+ }
ast_mutex_unlock(&p->lock);
}
break;
@@ -4217,10 +4217,6 @@
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)
@@ -5256,6 +5252,7 @@
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 */
+ p->t38.direct = 1;
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>");
}
@@ -6506,106 +6503,6 @@
}
}
-/*! \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;
- 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);
- 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(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(udptldest.sin_addr));
- snprintf(s, sizeof(s), "s=session\r\n");
- snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(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);
- if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL)
- ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxFillBitRemoval\r\n");
- if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR)
- ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingMMR\r\n");
- if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG)
- ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingJBIG\r\n");
- 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");
- len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + 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,
@@ -6635,7 +6532,7 @@
#define SDP_SAMPLE_RATE(x) 8000
/*! \brief Add Session Description Protocol message */
-static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
+static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int add_audio, int add_t38)
{
int len = 0;
int alreadysent = 0;
@@ -6655,26 +6552,33 @@
char *hold;
char m_audio[256]; /* Media declaration line for audio */
char m_video[256]; /* Media declaration line for video */
+ char m_modem[256]; /* Media declaration line for t38 */
char a_audio[1024]; /* Attributes for audio */
char a_video[1024]; /* Attributes for video */
+ char a_modem[1024]; /* Attributes for t38 */
char *m_audio_next = m_audio;
char *m_video_next = m_video;
+ char *m_modem_next = m_modem;
size_t m_audio_left = sizeof(m_audio);
size_t m_video_left = sizeof(m_video);
+ size_t m_modem_left = sizeof(m_modem);
char *a_audio_next = a_audio;
char *a_video_next = a_video;
+ char *a_modem_next = a_modem;
size_t a_audio_left = sizeof(a_audio);
size_t a_video_left = sizeof(a_video);
+ size_t a_modem_left = sizeof(a_modem);
int x;
- int capability;
+ int capability = 0;
int needvideo = FALSE;
int debug = sip_debug_test_pvt(p);
int min_audio_packet_size = 0;
int min_video_packet_size = 0;
m_video[0] = '\0'; /* Reset the video media string if it's not needed */
-
+ m_modem[0] = '\0';
+
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return AST_FAILURE;
@@ -6701,64 +6605,8 @@
dest.sin_port = sin.sin_port;
}
- capability = p->jointcapability;
-
-
- if (option_debug > 1) {
- char codecbuf[SIPBUFSIZE];
- 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));
- }
-
-#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
- 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);
- }
-#endif
-
- /* Check if we need video in this call */
- if ((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
- if (p->vrtp) {
- needvideo = TRUE;
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "This call needs video offers!\n");
- } else if (option_debug > 1)
- ast_log(LOG_DEBUG, "This call needs video offers, but there's no video support enabled!\n");
- }
-
-
- /* 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) {
- /* Determine video destination */
- if (p->vredirip.sin_addr.s_addr) {
- vdest.sin_addr = p->vredirip.sin_addr;
- vdest.sin_port = p->vredirip.sin_port;
- } else {
- vdest.sin_addr = p->ourip;
- vdest.sin_port = vsin.sin_port;
- }
- ast_build_string(&m_video_next, &m_video_left, "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), ntohs(vsin.sin_port));
- }
-
- if (debug)
- ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(sin.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=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
+ snprintf(owner, sizeof(owner), "o=root %d %d IN IP4 %s\r\n", 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_build_string(&m_audio_next, &m_audio_left, "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";
@@ -6767,98 +6615,201 @@
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, SDP_SAMPLE_RATE(codec),
- &m_audio_next, &m_audio_left,
- &a_audio_next, &a_audio_left,
- debug, &min_audio_packet_size);
- alreadysent |= codec;
- }
-
- /* Start by sending our preferred audio 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, SDP_SAMPLE_RATE(codec),
- &m_audio_next, &m_audio_left,
- &a_audio_next, &a_audio_left,
- debug, &min_audio_packet_size);
- alreadysent |= codec;
- }
-
- /* Now send any other common audio and video codecs, and non-codec formats: */
- for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
- if (!(capability & x)) /* Codec not requested */
- continue;
-
- if (alreadysent & x) /* Already added to SDP */
- continue;
-
- if (x <= AST_FORMAT_MAX_AUDIO)
- add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
+ if (add_audio) {
+ capability = p->jointcapability;
+
+
+ if (option_debug > 1) {
+ char codecbuf[SIPBUFSIZE];
+ 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));
+ }
+
+#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
+ 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);
+ }
+#endif
+
+ /* Check if we need video in this call */
+ if ((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
+ if (p->vrtp) {
+ needvideo = TRUE;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "This call needs video offers!\n");
+ } else if (option_debug > 1)
+ ast_log(LOG_DEBUG, "This call needs video offers, but there's no video support enabled!\n");
+ }
+
+
+ /* 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) {
+ /* Determine video destination */
+ if (p->vredirip.sin_addr.s_addr) {
+ vdest.sin_addr = p->vredirip.sin_addr;
+ vdest.sin_port = p->vredirip.sin_port;
+ } else {
+ vdest.sin_addr = p->ourip;
+ vdest.sin_port = vsin.sin_port;
+ }
+ ast_build_string(&m_video_next, &m_video_left, "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), ntohs(vsin.sin_port));
+ }
+
+ if (debug)
+ ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(sin.sin_port));
+
+ ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
+
+ /* 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, SDP_SAMPLE_RATE(codec),
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug, &min_audio_packet_size);
- else
- add_codec_to_sdp(p, x, 90000,
- &m_video_next, &m_video_left,
- &a_video_next, &a_video_left,
- debug, &min_video_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, 8000,
- &m_audio_next, &m_audio_left,
- &a_audio_next, &a_audio_left,
- debug);
- }
-
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n");
-
- if (!p->owner || !ast_internal_timing_enabled(p->owner))
- ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");
-
- if (min_audio_packet_size)
- ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size);
-
- if (min_video_packet_size)
- ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size);
-
- if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0))
- ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
-
- ast_build_string(&m_audio_next, &m_audio_left, "\r\n");
- if (needvideo)
- ast_build_string(&m_video_next, &m_video_left, "\r\n");
-
- len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
+ alreadysent |= codec;
+ }
+
+ /* Start by sending our preferred audio 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, SDP_SAMPLE_RATE(codec),
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug, &min_audio_packet_size);
+ alreadysent |= codec;
+ }
+
+ /* Now send any other common audio and video codecs, and non-codec formats: */
+ for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
+ if (!(capability & x)) /* Codec not requested */
+ continue;
+
+ if (alreadysent & x) /* Already added to SDP */
+ continue;
+
+ if (x <= AST_FORMAT_MAX_AUDIO)
+ add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug, &min_audio_packet_size);
+ else
+ add_codec_to_sdp(p, x, 90000,
+ &m_video_next, &m_video_left,
+ &a_video_next, &a_video_left,
+ debug, &min_video_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, 8000,
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug);
+ }
+
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n");
+
+ if (!p->owner || !ast_internal_timing_enabled(p->owner))
+ ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");
+
+ if (min_audio_packet_size)
+ ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size);
+
+ if (min_video_packet_size)
+ ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size);
+
+ if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0))
+ ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
+
+ ast_build_string(&m_audio_next, &m_audio_left, "\r\n");
+ if (needvideo)
+ ast_build_string(&m_video_next, &m_video_left, "\r\n");
+ }
+
+ if (add_t38 && p->udptl) {
+ struct sockaddr_in udptlsin;
+ struct sockaddr_in udptldest = { 0, };
+
+ ast_udptl_get_us(p->udptl, &udptlsin);
+
+ 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(p->ourip), ntohs(udptlsin.sin_port));
+ 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);
+ }
+
+ 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);
+ if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL)
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxFillBitRemoval\r\n");
+ if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR)
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingMMR\r\n");
+ if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG)
+ ast_build_string(&a_modem_next, &a_modem_left, "a=T38FaxTranscodingJBIG\r\n");
+ 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");
+ }
+
+ len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime);
+ if (add_audio)
+ len += strlen(m_audio) + strlen(a_audio) + strlen(hold);
if (needvideo) /* only if video response is appropriate */
len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold);
+ if (add_t38) {
+ len += strlen(m_modem) + strlen(a_modem);
+ }
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
@@ -6869,13 +6820,19 @@
if (needvideo) /* only if video response is appropriate */
add_line(resp, bandwidth);
add_line(resp, stime);
- add_line(resp, m_audio);
- add_line(resp, a_audio);
- add_line(resp, hold);
+ if (add_audio) {
+ add_line(resp, m_audio);
+ add_line(resp, a_audio);
+ add_line(resp, hold);
+ }
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video);
add_line(resp, a_video);
add_line(resp, hold); /* Repeat hold for the video stream */
+ }
+ if (add_t38) {
+ add_line(resp, m_modem);
+ add_line(resp, a_modem);
}
/* Update lastrtprx when we send our SDP */
@@ -6901,8 +6858,7 @@
}
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, 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)
@@ -6945,8 +6901,13 @@
ast_log(LOG_DEBUG, "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);
+ try_suggested_sip_codec(p);
+ if (p->t38.state == T38_PEER_DIRECT || p->t38.state == T38_ENABLED) {
+ p->t38.state = T38_ENABLED;
+ add_sdp(&resp, p, 1, 1);
+ } else {
+ add_sdp(&resp, p, 1, 0);
+ }
} 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)
@@ -7013,7 +6974,7 @@
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent");
- add_sdp(&req, p);
+ add_sdp(&req, p, 1, 0);
/* Use this as the basis */
initialize_initreq(p, &req);
p->lastinvite = p->ocseq;
@@ -7035,8 +6996,8 @@
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);
+ add_sdp(&req, p, 0, 1);
+
/* Use this as the basis */
initialize_initreq(p, &req);
ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */
@@ -7362,13 +7323,13 @@
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);
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);
+ add_sdp(&req, p, 0, 1);
} else if (p->rtp)
- add_sdp(&req, p);
+ add_sdp(&req, p, 1, 0);
} else {
add_header_contentLength(&req, 0);
}
@@ -12506,11 +12467,6 @@
ast_rtp_set_rtptimers_onhold(p->rtp);
if (p->vrtp)
ast_rtp_set_rtptimers_onhold(p->vrtp); /* Turn off RTP timers while we send fax */
- } else if (p->t38.state == T38_DISABLED && bridgepeer && (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 */
- /* XXXX Should we really destroy this session here, without any response at all??? */
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
} else {
if (option_debug > 1)
@@ -12533,7 +12489,7 @@
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 (p->t38.state == T38_LOCAL_REINVITE) {
/* 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)
@@ -12643,21 +12599,7 @@
/* While figuring that out, hangup the call */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- } 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.
- */
- p->t38.state = T38_DISABLED;
- /* 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");
-
- /* The dialog is now terminated */
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- sip_alreadygone(p);
} else {
/* We can't set up this call, so give up */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
@@ -14817,34 +14759,9 @@
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 */
- int sendok = TRUE;
-
- /* 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 ((bridgepeer->tech == &sip_tech || bridgepeer->tech == &sip_tech_info) && !ast_check_hangup(bridgepeer)) {
- bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
- /* Does the bridged peer have T38 ? */
- 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);
- sendok = FALSE;
- }
- /* No bridged peer with T38 enabled*/
- }
- }
- /* Respond to normal re-invite */
- if (sendok) {
- /* If this is not a re-invite or something to ignore - it's critical */
- ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (ast_test_flag(req, SIP_PKT_IGNORE) ? XMIT_UNRELIABLE : XMIT_CRITICAL)));
- }
+ /* If this is not a re-invite or something to ignore - it's critical */
+ ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+ transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (ast_test_flag(req, SIP_PKT_IGNORE) ? XMIT_UNRELIABLE : XMIT_CRITICAL)));
}
p->invitestate = INV_TERMINATED;
break;
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