[asterisk-commits] rizzo: branch rizzo/astobj2 r57961 -
/team/rizzo/astobj2/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Mar 5 14:49:49 MST 2007
Author: rizzo
Date: Mon Mar 5 15:49:48 2007
New Revision: 57961
URL: http://svn.digium.com/view/asterisk?view=rev&rev=57961
Log:
partial merge of svn 54838, missing block at line 6608 and below
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=57961&r1=57960&r2=57961
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Mon Mar 5 15:49:48 2007
@@ -486,6 +486,7 @@
#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
+#define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_ALLOW_EXT_DOM TRUE
#define DEFAULT_REALM "asterisk"
#define DEFAULT_NOTIFYRINGING TRUE
@@ -554,6 +555,7 @@
static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
+static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
static int compactheaders; /*!< send compact sip headers */
static int recordhistory; /*!< Record SIP history. Off by default */
static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
@@ -825,10 +827,14 @@
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (1 << 24) /*!< 24: Inactive */
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< 25: ???? */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< 26: Buggy CISCO MWI fix */
+#define SIP_PAGE2_NOTEXT (1 << 27) /*!< 26: Text not supported */
+#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< 27: Global text enable */
+#define SIP_PAGE2_DEBUG_TEXT (1 << 29) /*!< 28: Global text debug */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
- SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI)
+ SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
+ SIP_PAGE2_TEXTSUPPORT )
/* T.38 set of flags */
#define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
@@ -859,6 +865,7 @@
#define sipdebug ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
#define sipdebug_config ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
#define sipdebug_console ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
+#define sipdebug_text ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_TEXT)
/*! \brief T38 States for a call */
enum t38state {
@@ -1008,6 +1015,7 @@
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
+ struct sockaddr_in tredirip; /*!< Where our Text RTP should be going if not to us */
time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */
@@ -1036,6 +1044,7 @@
struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
struct ast_rtp *rtp; /*!< RTP Session */
struct ast_rtp *vrtp; /*!< Video RTP session */
+ struct ast_rtp *trtp; /*!< Text RTP session */
struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
struct sip_history_head *history; /*!< History of this SIP dialog */
struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
@@ -1677,9 +1686,10 @@
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
/*----- RTP interface functions */
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
+tatic enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static int sip_get_codec(struct ast_channel *chan);
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
@@ -1708,6 +1718,7 @@
.read = sip_read, /* called with chan locked */
.write = sip_write, /* called with chan locked */
.write_video = sip_write, /* called with chan locked */
+ .write_text = sip_write, /* XXX */
.indicate = sip_indicate, /* called with chan locked */
.transfer = sip_transfer, /* called with chan locked */
.fixup = sip_fixup, /* called with chan locked */
@@ -1731,6 +1742,7 @@
.read = sip_read, /* called with chan locked */
.write = sip_write, /* called with chan locked */
.write_video = sip_write, /* called with chan locked */
+ .write_text = sip_write, /* XXX */
.indicate = sip_indicate, /* called with chan locked */
.transfer = sip_transfer, /* called with chan locked */
.fixup = sip_fixup, /* called with chan locked */
@@ -1778,6 +1790,7 @@
type: "SIP",
get_rtp_info: sip_get_rtp_peer,
get_vrtp_info: sip_get_vrtp_peer,
+ get_trtp_info: sip_get_trtp_peer,
set_rtp_peer: sip_set_rtp_peer,
get_codec: sip_get_codec,
};
@@ -3081,6 +3094,11 @@
ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
ast_udptl_setnat(p->udptl, natflags);
}
+ if (p->trtp) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting NAT on TRTP to %s\n", mode);
+ ast_rtp_setnat(p->trtp, natflags);
+ }
}
/*! \brief Create address structure from peer reference.
@@ -3101,6 +3119,10 @@
if ((!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(dialog->capability & AST_FORMAT_VIDEO_MASK)) && dialog->vrtp) {
ast_rtp_destroy(dialog->vrtp);
dialog->vrtp = NULL;
+ }
+ if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT) && dialog->trtp) {
+ ast_rtp_destroy(dialog->trtp);
+ dialog->trtp = NULL;
}
dialog->prefs = peer->prefs;
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
@@ -3139,6 +3161,13 @@
ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
+ }
+ if (dialog->trtp) {
+ ast_rtp_setdtmf(dialog->trtp, 0);
+ ast_rtp_setdtmfcompensate(dialog->trtp, 0);
+ ast_rtp_set_rtptimeout(dialog->trtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->trtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->trtp, peer->rtpkeepalive);
}
ast_string_field_set(dialog, peername, peer->username);
@@ -3427,6 +3456,8 @@
ast_rtp_destroy(p->rtp);
if (p->vrtp)
ast_rtp_destroy(p->vrtp);
+ if (p->trtp)
+ ast_rtp_destroy(p->trtp);
if (p->udptl)
ast_udptl_destroy(p->udptl);
sip_refer_destroy(p);
@@ -3908,10 +3939,13 @@
if (!p->pendinginvite) {
char *audioqos = "";
char *videoqos = "";
+ char *textqos = "";
if (p->rtp)
audioqos = ast_rtp_get_quality(p->rtp);
if (p->vrtp)
videoqos = ast_rtp_get_quality(p->vrtp);
+ if (p->trtp)
+ textqos = ast_rtp_get_quality(p->trtp);
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
@@ -3921,11 +3955,15 @@
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
+ if (p->trtp)
+ append_history(p, "RTCPtext", "Quality:%s", textqos);
}
if (p->rtp && oldowner)
pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
if (p->vrtp && oldowner)
pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
+ if (p->trtp && oldowner)
+ pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", textqos);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@@ -4043,6 +4081,23 @@
sip_pvt_unlock(p);
}
break;
+ case AST_FRAME_TEXT:
+ if (p) {
+ sip_pvt_lock(p);
+ if (p->trtp) {
+ /* Activate text early media */
+ if ((ast->_state != AST_STATE_UP) &&
+ !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
+ !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ }
+ p->lastrtptx = time(NULL);
+ res = ast_rtp_write(p->trtp, frame);
+ }
+ sip_pvt_unlock(p);
+ }
+ break;
case AST_FRAME_IMAGE:
return 0;
break;
@@ -4274,7 +4329,10 @@
struct ast_variable *v = NULL;
int fmt;
int what;
+ int video;
+ int text;
int needvideo = 0;
+ int needtext = 0;
{
const char *my_name; /* pick a good name */
@@ -4300,15 +4358,22 @@
/* Select our native format based on codec preference until we receive
something from another device to the contrary. */
- if (i->jointcapability) /* The joint capabilities of us and peer */
+ if (i->jointcapability) { /* The joint capabilities of us and peer */
what = i->jointcapability;
- else if (i->capability) /* Our configured capability for this peer */
+ video = i->jointcapability & AST_FORMAT_VIDEO_MASK;
+ text = i->jointcapability & AST_FORMAT_TEXT_MASK;
+ } else if (i->capability) { /* Our configured capability for this peer */
what = i->capability;
- else
+ video = i->capability & AST_FORMAT_VIDEO_MASK;
+ text = i->capability & AST_FORMAT_TEXT_MASK;
+ } else {
what = global_capability; /* Global codec support */
+ video = global_capability & AST_FORMAT_VIDEO_MASK;
+ text = global_capability & AST_FORMAT_TEXT_MASK;
+ }
/* Set the native formats for audio and merge in video */
- tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
+ tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text;
if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
@@ -4333,6 +4398,13 @@
needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */
}
+ if (i->trtp) {
+ if (i->prefcodec)
+ needtext = i->prefcodec & AST_FORMAT_TEXT_MASK; /* Outbound call */
+ else
+ needtext = i->jointcapability & AST_FORMAT_TEXT_MASK; /* Inbound call */
+ }
+
if (option_debug > 2) {
if (needvideo)
ast_log(LOG_DEBUG, "This channel can handle video! HOLLYWOOD next!\n");
@@ -4355,6 +4427,9 @@
if (needvideo && i->vrtp) {
tmp->fds[2] = ast_rtp_fd(i->vrtp);
tmp->fds[3] = ast_rtcp_fd(i->vrtp);
+ }
+ if (needtext && i->trtp) {
+ tmp->fds[4] = ast_rtp_fd(i->trtp);
}
if (i->udptl) {
tmp->fds[5] = ast_udptl_fd(i->udptl);
@@ -4575,6 +4650,19 @@
case 3:
f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */
break;
+ case 4:
+ f = ast_rtp_read(p->trtp); /* RTP Text */
+ if (sipdebug_text) {
+ int i;
+ unsigned char* arr = f->data;
+ for (i=0; i < f->datalen; i++)
+ ast_verbose("%c", (arr[i] > ' ' && arr[i] < '}') ? arr[i] : '.');
+ ast_verbose(" -> ");
+ for (i=0; i < f->datalen; i++)
+ ast_verbose("%02X ", arr[i]);
+ ast_verbose("\n");
+ }
+ break;
case 5:
f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
break;
@@ -4592,7 +4680,7 @@
if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
if (option_debug)
ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
- p->owner->nativeformats = (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass;
+ p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK) ) | f->subclass;
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
@@ -4766,11 +4854,15 @@
/* If the global videosupport flag is on, we always create a RTP interface for video */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ f (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT))
+ p->trtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
- if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) {
- ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n",
- ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno));
+ f (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
+ || (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
+ ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
+ ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",
+ ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno));
return __sip_destroy(p);
}
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
@@ -4786,6 +4878,11 @@
ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
+ }
+ if (p->trtp) {
+ ast_rtp_settos(p->trtp, global_tos_text);
+ ast_rtp_setdtmf(p->trtp, 0);
+ ast_rtp_setdtmfcompensate(p->trtp, 0);
}
if (p->udptl)
ast_udptl_settos(p->udptl, global_tos_audio);
@@ -5299,6 +5396,7 @@
int len = -1;
int portno = -1; /*!< RTP Audio port number */
int vportno = -1; /*!< RTP Video port number */
+ int tportno = -1; /*!< RTP Text port number */
int udptlportno = -1;
int peert38capability = 0;
char s[256];
@@ -5307,20 +5405,24 @@
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
int peercapability = 0, peernoncodeccapability = 0;
int vpeercapability = 0, vpeernoncodeccapability = 0;
+ int tpeercapability = 0, tpeernoncodeccapability = 0;
struct sockaddr_in sin; /*!< media socket address */
struct sockaddr_in vsin; /*!< Video socket address */
+ struct sockaddr_in tsin; /*!< Text socket address */
const char *codecs;
struct hostent *hp; /*!< RTP Audio host IP */
struct hostent *vhp = NULL; /*!< RTP video host IP */
+ struct hostent *thp = NULL; /*!< RTP text host IP */
struct ast_hostent audiohp;
struct ast_hostent videohp;
+ struct ast_hostent texthp;
int codec;
int destiterator = 0;
int iterator;
int sendonly = 0;
int numberofports;
- struct ast_rtp *newaudiortp, *newvideortp; /* Buffers for codec handling */
+ struct ast_rtp *newaudiortp, *newvideortp, *newtextrtp; /* Buffers for codec handling */
int newjointcapability; /* Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
@@ -5346,6 +5448,11 @@
ast_rtp_new_init(newvideortp);
ast_rtp_pt_clear(newvideortp);
+ newtextrtp = alloca(ast_rtp_alloc_size());
+ memset(newtextrtp, 0, ast_rtp_alloc_size());
+ ast_rtp_new_init(newtextrtp);
+ ast_rtp_pt_clear(newtextrtp);
+
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -5372,15 +5479,24 @@
return -1;
}
vhp = hp; /* Copy to video address as default too */
+ thp = hp; /* Copy to video address as default too */
iterator = req->sdp_start;
ast_set_flag(&p->flags[0], SIP_NOVIDEO);
-
+ ast_set_flag(&p->flags[1], SIP_PAGE2_NOTEXT);
+
+ if (p->vrtp)
+ ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
+
+ if (p->trtp)
+ ast_rtp_pt_clear(newtextrtp); /* Must be cleared in case no m=text line exists */
/* Find media streams in this SDP offer */
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
int x;
int audio = FALSE;
+ int video = FALSE;
+ int text = FALSE;
numberofports = 1;
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
@@ -5401,7 +5517,7 @@
}
} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
- /* If it is not audio - is it video ? */
+ video = TRUE;
ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
numberofmediastreams++;
vportno = x;
@@ -5414,6 +5530,22 @@
if (debug)
ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_set_m_type(newvideortp, codec);
+ }
+ } else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
+ (sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1)) {
+ text = TRUE;
+ ast_clear_flag(&p->flags[1], SIP_PAGE2_NOTEXT);
+ numberofmediastreams++;
+ tportno = x;
+ /* Scan through the RTP payload types specified in a "m=" line: */
+ for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
+ if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
+ ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
+ return -1;
+ }
+ if (debug)
+ ast_verbose("Found RTP text format %d\n", codec);
+ ast_rtp_set_m_type(newtextrtp, codec);
}
} else if (p->udptl && ( (sscanf(m, "image %d udptl t38%n", &x, &len) == 1) ||
(sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1) )) {
@@ -5447,27 +5579,35 @@
if (audio) {
if ( !(hp = ast_gethostbyname(host, &audiohp)))
ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
- } else if (!(vhp = ast_gethostbyname(host, &videohp)))
- ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
+ } else if (video) {
+ if (!(vhp = ast_gethostbyname(host, &videohp)))
+ ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
+ } else if (text) {
+ if (!(thp = ast_gethostbyname(host, &texthp)))
+ ast_log(LOG_WARNING, "Unable to lookup RTP text host in secondary c= line, '%s'\n", c);
+ }
}
}
}
- if (portno == -1 && vportno == -1 && udptlportno == -1)
+ if (portno == -1 && vportno == -1 && udptlportno == -1 && tportno == -1)
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
return -2;
- if (numberofmediastreams > 2)
- /* We have too many fax, audio and/or video media streams, fail this offer */
+ if (numberofmediastreams > 3)
+ /* We have too many fax, audio and/or video and/or text media streams, fail this offer */
return -3;
/* RTP addresses and ports for audio and video */
sin.sin_family = AF_INET;
vsin.sin_family = AF_INET;
+ tsin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
if (vhp)
memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
+ if (thp)
+ memcpy(&tsin.sin_addr, thp->h_addr, sizeof(tsin.sin_addr));
/* Setup UDPTL port number */
if (p->udptl) {
@@ -5501,11 +5641,15 @@
}
}
}
- /* Setup video port number */
+ /* Setup video port number, assumes we have audio */
if (vportno != -1)
vsin.sin_port = htons(vportno);
- /* Next, scan through each "a=rtpmap:" line, noting each
+ /* Setup text port number, assumes we have audio */
+ if (tportno != -1)
+ tsin.sin_port = htons(tportno);
+
+ /* Next, scan through each "a=xxxx:" line, noting each
* specified RTP payload type (with corresponding MIME subtype):
*/
/* XXX This needs to be done per media stream, since it's media stream specific */
@@ -5595,10 +5739,18 @@
last_rtpmap_codec++;
/* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
+ /* Note: This should all be done in the context of the m= above */
+ if (!strncasecmp(mimeSubtype, "H26",3)) { /* Video */
+ /* Not going to do anything here for the moment, but we will soon */
+ } else if (!strncasecmp(mimeSubtype, "T140",4)) { /* Text */
+ if (p->trtp) {
+ /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
+ ast_rtp_set_rtpmap_type(newtextrtp, codec, "text", mimeSubtype, 0);
+ }
+ } else { /* Must be audio?? */
+ ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0);
- if (p->vrtp)
- ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
+ }
}
}
@@ -5715,21 +5867,23 @@
/* Now gather all of the codecs that we are asked for: */
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
-
- newjointcapability = p->capability & (peercapability | vpeercapability);
- newpeercapability = (peercapability | vpeercapability);
+ ast_rtp_get_current_formats(newtextrtp, &tpeercapability, &tpeernoncodeccapability);
+
+ newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability);
+ newpeercapability = (peercapability | vpeercapability | tpeercapability);
newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
if (debug) {
/* shame on whoever coded this.... */
- char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ];
-
- ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
+ char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ], s5[BUFSIZ];
+
+ ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
ast_getformatname_multiple(s1, BUFSIZ, p->capability),
ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
- ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
+ ast_getformatname_multiple(s4, BUFSIZ, tpeercapability),
+ ast_getformatname_multiple(s5, BUFSIZ, newjointcapability));
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
@@ -5758,6 +5912,8 @@
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
ast_rtp_pt_copy(p->vrtp, newvideortp);
+ if (p->trtp)
+ ast_rtp_pt_copy(p->trtp, newtextrtp);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -5785,6 +5941,12 @@
if (debug)
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
}
+ /* Setup text port number */
+ if (p->trtp && tsin.sin_port) {
+ ast_rtp_set_peer(p->trtp, &tsin);
+ if (debug)
+ ast_verbose("Peer text RTP is at port %s:%d\n", ast_inet_ntoa(tsin.sin_addr), ntohs(tsin.sin_port));
+ }
/* Ok, we're going with this offer */
if (option_debug > 1) {
@@ -5805,7 +5967,7 @@
ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
}
- p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
+ p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability);
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
@@ -6535,6 +6697,50 @@
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
*min_packet_size = fmt.cur_ms;
+/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
+ /* This is different to the audio one now so we can add more caps later */
+ static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+ char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
+ int debug, int *min_packet_size)
+ {
+ int rtp_code;
+
+ if (!p->vrtp)
+ return;
+
+ if (debug)
+ ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+
+ if ((rtp_code = ast_rtp_lookup_code(p->vrtp, 1, codec)) == -1)
+ return;
+
+ ast_build_string(m_buf, m_size, " %d", rtp_code);
+ ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
+ ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ /* Add fmtp code here */
+ }
+
+ /*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
+ static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+ char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
+ int debug, int *min_packet_size)
+ {
+ int rtp_code;
+
+ if (!p->trtp)
+ return;
+
+ if (debug)
+ ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+
+ if ((rtp_code = ast_rtp_lookup_code(p->trtp, 1, codec)) == -1)
+ return;
+
+ ast_build_string(m_buf, m_size, " %d", rtp_code);
+ ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
+ ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ /* Add fmtp code here */
+ }
/* Our first codec packetization processed cannot be zero */
if ((*min_packet_size)==0 && fmt.cur_ms)
*min_packet_size = fmt.cur_ms;
@@ -6684,12 +6890,14 @@
/*! \brief Set all IP media addresses for this call
\note called from add_sdp()
*/
-static void get_our_media_address(struct sip_pvt *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *dest, struct sockaddr_in *vdest)
+static void get_our_media_address(struct sip_pvt *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *tsin, struct sockaddr_in *dest, struct sockaddr_in *vdest)
{
/* First, get our address */
ast_rtp_get_us(p->rtp, sin);
if (p->vrtp)
ast_rtp_get_us(p->vrtp, vsin);
+ if (p->trtp)
+ ast_rtp_get_us(p->trtp, tsin);
/* Now, try to figure out where we want them to send data */
/* Is this a re-invite to move the media out, then use the original offer from caller */
@@ -6723,8 +6931,10 @@
struct sockaddr_in sin;
struct sockaddr_in vsin;
+ struct sockaddr_in tsin;
struct sockaddr_in dest;
struct sockaddr_in vdest = { 0, };
+ struct sockaddr_in tdest = { 0, };
/* SDP fields */
char *version = "v=0\r\n"; /* Protocol version */
@@ -17665,7 +17875,7 @@
}
/*! \brief Set the RTP peer for this call */
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
{
struct sip_pvt *p = chan->tech_pvt;
int changed = 0;
@@ -17700,6 +17910,12 @@
changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
} else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
memset(&p->vredirip, 0, sizeof(p->vredirip));
+ changed = 1;
+ }
+ if (trtp) {
+ changed |= ast_rtp_get_peer(trtp, &p->tredirip);
+ } else if (p->tredirip.sin_addr.s_addr || ntohs(p->tredirip.sin_port) != 0) {
+ memset(&p->tredirip, 0, sizeof(p->tredirip));
changed = 1;
}
if (codecs && (p->redircodecs != codecs)) {
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