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Merged revisions 184948 via svnmerge from

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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/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Joshua Colp 16 years ago
parent 87be216c37
commit 1c3ca72745
1 changed files with 237 additions and 268 deletions
Show Stats Download Patch File Download Diff File

505
channels/chan_sip.c
Unescape View File

@ -1056,7 +1056,6 @@ 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 */
@ -1070,6 +1069,7 @@ 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 */
@ -1798,7 +1798,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
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);
@ -3219,7 +3219,6 @@ 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:
@ -3960,6 +3959,10 @@ 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>");
@ -4303,9 +4306,6 @@ 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);
}
}
@ -5077,11 +5077,9 @@ static int sip_answer(struct ast_channel *ast)
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);
} else {
ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
}
ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
}
sip_pvt_unlock(p);
return res;
@ -5118,9 +5116,13 @@ 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);
}
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->rtp, frame);
}
sip_pvt_unlock(p);
}
@ -5171,8 +5173,16 @@ 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)
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;
@ -5589,10 +5599,6 @@ 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)
pbx_builtin_setvar_helper(tmp, v->name, v->value);
@ -7099,7 +7105,7 @@ 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) {
@ -8176,92 +8182,6 @@ 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, int sample_rate,
struct ast_str **m_buf, struct ast_str **a_buf,
@ -8331,7 +8251,7 @@ 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;
@ -8340,8 +8260,10 @@ 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 */
@ -8350,16 +8272,18 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
char connection[256]; /* Connection data */
char *stime = "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;
@ -8390,183 +8314,232 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
p->sessionversion++;
}
capability = p->jointcapability;
get_our_media_address(p, needvideo, &sin, &vsin, &tsin, &dest, &vdest);
/* 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));
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 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");
}
/* 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);
if (debug)
ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(sin.sin_port));
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));
/* 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));
}
/* 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) {
/* Check if we need text in this call */
if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
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;
} else {
tdest.sin_addr = p->ourip.sin_addr;
tdest.sin_port = tsin.sin_port;
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");
}
ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
/* 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;
} 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));
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 */
/* 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 */
/* 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));
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";
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
*/
/* 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;
/* 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, &a_audio,
debug, &min_audio_packet_size);
alreadysent |= codec;
}
add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(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;
/* 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 (!(codec = ast_codec_pref_index(&p->prefs, x)))
break;
if (!(capability & codec))
continue;
if (!(capability & codec))
continue;
if (alreadysent & codec)
if (alreadysent & codec)
continue;
add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
&m_audio, &a_audio,
debug, &min_audio_packet_size);
alreadysent |= codec;
}
add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(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;
/* 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 (alreadysent & x) /* Already added to SDP */
continue;
if (x & AST_FORMAT_AUDIO_MASK)
add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
&m_audio, &a_audio, debug, &min_audio_packet_size);
else if (x & AST_FORMAT_VIDEO_MASK)
add_vcodec_to_sdp(p, x, 90000,
&m_video, &a_video, debug, &min_video_packet_size);
else if (x & AST_FORMAT_TEXT_MASK)
add_tcodec_to_sdp(p, x, 1000,
&m_text, &a_text, debug, &min_text_packet_size);
}
if (x & AST_FORMAT_AUDIO_MASK)
add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
&m_audio, &a_audio, debug, &min_audio_packet_size);
else if (x & AST_FORMAT_VIDEO_MASK)
add_vcodec_to_sdp(p, x, 90000,
&m_video, &a_video, debug, &min_video_packet_size);
else if (x & AST_FORMAT_TEXT_MASK)
add_tcodec_to_sdp(p, x, 1000,
&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;
/* 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, &a_audio, debug);
}
ast_debug(3, "-- Done with adding codecs to SDP\n");
add_noncodec_to_sdp(p, x, 8000, &m_audio, &a_audio, debug);
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);
}
ast_debug(3, "-- Done with adding codecs to SDP\n");
if (add_t38) {
ast_udptl_get_us(p->udptl, &udptlsin);
if (!p->owner || !ast_internal_timing_enabled(p->owner))
ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
/* 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 (min_audio_packet_size)
ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
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));
/* 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);
/* 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\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");
}
/* 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)
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 (needaudio)
@ -8584,6 +8557,8 @@ 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);
@ -8609,6 +8584,10 @@ 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 ? */
@ -8631,7 +8610,7 @@ 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)
@ -8673,8 +8652,12 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
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)
@ -8754,9 +8737,9 @@ 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);
@ -9139,12 +9122,12 @@ 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 {
add_header_contentLength(&req, 0);
}
@ -15223,20 +15206,6 @@ 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);
p->needdestroy = 1;
sip_alreadygone(p);
} else {
/* We can't set up this call, so give up */
if (p->owner && !req->ignore)

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