#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
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 */
#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 )
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
#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 {
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 */
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 */
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);
+static 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);
.read = sip_read,
.write = sip_write,
.write_video = sip_write,
+ .write_text = sip_write,
.indicate = sip_indicate,
.transfer = sip_transfer,
.fixup = sip_fixup,
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,
};
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.
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)) {
dialog->t38.capability = global_t38_capability;
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);
ast_string_field_set(dialog, authname, peer->username);
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);
if (p->refer)
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);
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. */
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;
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 */
/* 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));
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");
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);
}
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;
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);
}
/* 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);
+ if (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));
+ if (!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));
ast_mutex_destroy(&p->pvt_lock);
if (p->chanvars) {
ast_variables_destroy(p->chanvars);
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);
p->maxcallbitrate = default_maxcallbitrate;
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];
/* 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;
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 ? */
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) ||
}
} 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;
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) )) {
if (debug)
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) {
}
}
}
- /* 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 */
last_rtpmap_codec++;
/* Note: should really look at the 'freq' and '#chans' params too */
- 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);
+ /* 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);
+ }
}
}
/* 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];
+ char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ], s5[BUFSIZ];
- ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
+ 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),
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);
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) {
char buf[BUFSIZ];
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);
}
*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 */
+}
+
+
/*! \brief Get Max T.38 Transmission rate from T38 capabilities */
static int t38_get_rate(int t38cap)
{
/*! \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 */
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 */
char *hold;
char m_audio[256]; /* Media declaration line for audio */
char m_video[256]; /* Media declaration line for video */
+ char m_text[256]; /* Media declaration line for text */
char a_audio[1024]; /* Attributes for audio */
char a_video[1024]; /* Attributes for video */
+ char a_text[1024]; /* Attributes for text */
char *m_audio_next = m_audio;
char *m_video_next = m_video;
+ char *m_text_next = m_text;
size_t m_audio_left = sizeof(m_audio);
size_t m_video_left = sizeof(m_video);
+ size_t m_text_left = sizeof(m_text);
char *a_audio_next = a_audio;
char *a_video_next = a_video;
+ char *a_text_next = a_text;
size_t a_audio_left = sizeof(a_audio);
size_t a_video_left = sizeof(a_video);
+ size_t a_text_left = sizeof(a_text);
int x;
int capability;
int needvideo = FALSE;
+ int needtext = FALSE;
int debug = sip_debug_test_pvt(p);
int min_audio_packet_size = 0;
int min_video_packet_size = 0;
+ int min_text_packet_size = 0;
m_video[0] = '\0'; /* Reset the video media string if it's not needed */
+ m_text[0] = '\0'; /* Reset the video media string if it's not needed */
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
if (option_debug > 1) {
char codecbuf[BUFSIZ];
- 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 capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
+ ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False");
ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
}
}
/* Get our media addresses */
- get_our_media_address(p, needvideo, &sin, &vsin, &dest, &vdest);
+ 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), 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_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));
+ /* Check if we need text in this call */
+ if((capability & AST_FORMAT_TEXT_MASK) && !ast_test_flag(&p->flags[1], SIP_PAGE2_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;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "This call needs text offers! \n");
+ } else if (option_debug > 1)
+ ast_log(LOG_DEBUG, "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;
+ tdest.sin_port = tsin.sin_port;
+ }
+ ast_build_string(&m_text_next, &m_text_left, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
+
+ if (debug)
+ ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(tsin.sin_port));
+
+ }
/* Start building generic SDP headers */
}
/* 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) {
+ for (x = 1; x <= (needtext ? AST_FORMAT_MAX_TEXT : (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO)); x <<= 1) {
if (!(capability & x)) /* Codec not requested */
continue;
&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,
+ else if (x <= AST_FORMAT_MAX_VIDEO)
+ add_vcodec_to_sdp(p, x, 90000,
&m_video_next, &m_video_left,
&a_video_next, &a_video_left,
debug, &min_video_packet_size);
+ else if (x <= AST_FORMAT_MAX_TEXT)
+ add_tcodec_to_sdp(p, x, 1000,
+ &m_text_next, &m_text_left,
+ &a_text_next, &a_text_left,
+ debug, &min_text_packet_size);
}
/* Now add DTMF RFC2833 telephony-event as a codec */
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);
- if (needvideo) /* only if video response is appropriate */
- len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold);
+ if (min_audio_packet_size)
+ ast_build_string(&a_audio_next, &a_audio_left, "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_build_string(&a_video_next, &a_video_left, "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_build_string(&a_text_next, &a_text_left, "a=ptime:%d\r\n", min_text_packet_size);
+
+ if ((m_audio_left < 2) || (m_video_left < 2) || (m_text_left < 2) ||
+ (a_audio_left == 0) || (a_video_left == 0) || (a_text_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 (needtext)
+ ast_build_string(&m_text_next, &m_text_left, "\r\n");
+
+ len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + 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 (needtext) /* only if text response is appropriate */
+ len += strlen(m_text) + strlen(a_text) + strlen(hold);
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
add_line(resp, a_video);
add_line(resp, hold); /* Repeat hold for the video stream */
}
+ if (needtext) { /* only if text response is appropriate */
+ add_line(resp, m_text);
+ add_line(resp, a_text);
+ add_line(resp, hold); /* Repeat hold for the text stream */
+ }
/* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
}
+ /* If we do not support text, remove text from call structure */
+ if (!ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && p->trtp) {
+ ast_rtp_destroy(p->trtp);
+ p->trtp = NULL;
+ }
}
unref_user(user);
return res;
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
}
+ if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) || !(p->capability & AST_FORMAT_TEXT_MASK)) && p->trtp) {
+ ast_rtp_destroy(p->trtp);
+ p->trtp = NULL;
+ }
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
"Dynamic: %s\r\n"
"Natsupport: %s\r\n"
"VideoSupport: %s\r\n"
+ "TextSupport: %s\r\n"
"ACL: %s\r\n"
"Status: %s\r\n"
"RealtimeDevice: %s\r\n\r\n",
ast_test_flag(&iterator->flags[1], SIP_PAGE2_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */
ast_test_flag(&iterator->flags[0], SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */
ast_test_flag(&iterator->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */
+ ast_test_flag(&iterator->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no", /* TEXTSUPPORT=yes? */
iterator->ha ? "yes" : "no", /* permit/deny */
status,
realtimepeers ? (ast_test_flag(&iterator->flags[0], SIP_REALTIME) ? "yes":"no") : "no");
ast_cli(fd, " PromiscRedir : %s\n", ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Yes":"No");
ast_cli(fd, " User=Phone : %s\n", ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Yes":"No");
ast_cli(fd, " Video Support: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Yes":"No");
+ ast_cli(fd, " Text Support : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Yes":"No");
ast_cli(fd, " Trust RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_TRUSTRPID) ? "Yes" : "No");
ast_cli(fd, " Send RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_SENDRPID) ? "Yes" : "No");
ast_cli(fd, " Subscriptions: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
+ astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port));
ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
ast_cli(fd, " Videosupport: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "Yes" : "No");
+ ast_cli(fd, " Textsupport: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT) ? "Yes" : "No");
ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No");
ast_cli(fd, " MatchAuthUsername: %s\n", global_match_auth_username ? "Yes" : "No");
ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No");
ast_cli(fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
ast_cli(fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
ast_cli(fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
+ ast_cli(fd, " IP ToS RTP text: %s\n", ast_tos2str(global_tos_text));
ast_cli(fd, " T38 fax pt UDPTL: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL) ? "Yes" : "No");
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
ast_cli(fd, " T38 fax pt RTP: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP) ? "Yes" : "No");
return RESULT_SHOWUSAGE;
}
ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE);
+ ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_TEXT); /*! \note this can be a special debug command - "sip debug text" or something */
memset(&debugaddr, 0, sizeof(debugaddr));
ast_cli(fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : "");
return RESULT_SUCCESS;
if (argc != 4)
return RESULT_SHOWUSAGE;
ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE);
+ ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_TEXT);
ast_cli(fd, "SIP Debugging Disabled\n");
return RESULT_SUCCESS;
}
ast_rtp_stop(p->rtp);
if (p->vrtp)
ast_rtp_stop(p->vrtp);
+ if (p->trtp)
+ ast_rtp_stop(p->trtp);
if (p->udptl)
ast_udptl_stop(p->udptl);
}
/* Get RTCP quality before end of call */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
- char *audioqos, *videoqos;
+ char *audioqos, *videoqos, *textqos;
if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp);
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
}
+ if (p->trtp) {
+ textqos = ast_rtp_get_quality(p->trtp);
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ append_history(p, "RTCPtext", "Quality:%s", textqos);
+ if (p->owner)
+ pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
+ }
}
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif
p->prefcodec = oldformat; /* Format for this call */
+ p->jointcapability = oldformat;
sip_pvt_lock(p);
tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */
sip_pvt_unlock(p);
ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT);
res = 1;
+ } else if (!strcasecmp(v->name, "textsupport")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_TEXTSUPPORT);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_TEXTSUPPORT);
+ res = 1;
} else if (!strcasecmp(v->name, "allowoverlap")) {
ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP);
global_tos_sip = DEFAULT_TOS_SIP;
global_tos_audio = DEFAULT_TOS_AUDIO;
global_tos_video = DEFAULT_TOS_VIDEO;
+ global_tos_text = DEFAULT_TOS_TEXT;
externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */
externexpire = 0; /* Expiration for DNS re-issuing */
externrefresh = 10;
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT);
+ ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
/* Read the [general] config section of sip.conf (or from realtime config) */
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
} else if (!strcasecmp(v->name, "tos_video")) {
if (ast_str2tos(v->value, &global_tos_video))
ast_log(LOG_WARNING, "Invalid tos_video value at line %d, recommended value is 'af41'. See doc/ip-tos.txt.\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_text")) {
+ if (ast_str2tos(v->value, &global_tos_text))
+ ast_log(LOG_WARNING, "Invalid tos_text value at line %d, recommended value is 'af41'. See doc/ip-tos.txt.\n", v->lineno);
} else if (!strcasecmp(v->name, "bindport")) {
if (sscanf(v->value, "%d", &ourport) == 1) {
bindaddr.sin_port = htons(ourport);
return res;
}
+/*! \brief Returns null if we can't reinvite text (part of RTP interface) */
+static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+{
+ struct sip_pvt *p = NULL;
+ enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+
+ if (!(p = chan->tech_pvt))
+ return AST_RTP_GET_FAILED;
+
+ sip_pvt_lock(p);
+ if (!(p->trtp)) {
+ sip_pvt_unlock(p);
+ return AST_RTP_GET_FAILED;
+ }
+
+ *rtp = p->trtp;
+
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ res = AST_RTP_TRY_NATIVE;
+
+ sip_pvt_unlock(p);
+
+ return res;
+}
+
/*! \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;
int changed = 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)) {
p->redircodecs = codecs;
changed = 1;
unsigned int lastrxts;
unsigned int lastividtimestamp;
unsigned int lastovidtimestamp;
+ unsigned int lastitexttimestamp;
+ unsigned int lastotexttimestamp;
unsigned int lasteventseqn;
int lastrxseqno; /*!< Last received sequence number */
unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
return f ? f : &ast_null_frame;
}
rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
- rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;
+ rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : (rtp->f.subclass < AST_FORMAT_MAX_VIDEO) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
if (!rtp->lastrxts)
rtp->lastrxts = timestamp;
rtp->f.ts = timestamp / 8;
rtp->f.len = rtp->f.samples / 8;
rtp->f.seqno = seqno;
- } else {
+ } else if(rtp->f.subclass < AST_FORMAT_MAX_VIDEO) {
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)
rtp->lastividtimestamp = timestamp;
rtp->f.delivery.tv_usec = 0;
if (mark)
rtp->f.subclass |= 0x1;
-
+ } else {
+ /* TEXT -- samples is # of samples vs. 1000 */
+ if (!rtp->lastitexttimestamp)
+ rtp->lastitexttimestamp = timestamp;
+ rtp->f.samples = timestamp - rtp->lastitexttimestamp;
+ rtp->lastitexttimestamp = timestamp;
+ rtp->f.delivery.tv_sec = 0;
+ rtp->f.delivery.tv_usec = 0;
}
rtp->f.src = "RTP";
return &rtp->f;
// dest = c0, src = c1
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
+ struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
- enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
- enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
+ enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
+ enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
int srccodec, nat_active = 0;
/* Lock channels */
return -1;
}
- /* Get audio and video interface (if native bridge is possible) */
+ /* Get audio, video and text interface (if native bridge is possible) */
audio_dest_res = destpr->get_rtp_info(c0, &destp);
video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED;
+ text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(c0, &tdestp) : AST_RTP_GET_FAILED;
if (srcpr) {
audio_src_res = srcpr->get_rtp_info(c1, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED;
+ text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
}
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (srcp && (srcp->nat || ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
nat_active = 1;
/* Bridge media early */
- if (destpr->set_rtp_peer(c0, srcp, vsrcp, srccodec, nat_active))
+ if (destpr->set_rtp_peer(c0, srcp, vsrcp, tsrcp, srccodec, nat_active))
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
ast_channel_unlock(c0);
if (c1)
{
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
+ struct ast_rtp *tdestp = NULL, *tsrcp = NULL; /* Text RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
- enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
- enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
+ enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED;
+ enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED;
int srccodec;
/* Lock channels */
/* Get audio and video interface (if native bridge is possible) */
audio_dest_res = destpr->get_rtp_info(dest, &destp);
video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
+ text_dest_res = destpr->get_trtp_info ? destpr->get_trtp_info(dest, &tdestp) : AST_RTP_GET_FAILED;
audio_src_res = srcpr->get_rtp_info(src, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
+ text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(src, &tsrcp) : AST_RTP_GET_FAILED;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE) {
ast_rtp_pt_copy(destp, srcp);
if (vdestp && vsrcp)
ast_rtp_pt_copy(vdestp, vsrcp);
+ if (tdestp && tsrcp)
+ ast_rtp_pt_copy(tdestp, tsrcp);
if (srcpr->get_codec)
srccodec = srcpr->get_codec(src);
else
srccodec = 0;
if (media) {
/* Bridge early */
- if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
+ if (destpr->set_rtp_peer(dest, srcp, vsrcp, tsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src->name);
}
ast_channel_unlock(dest);
rtp->lastrxts = 0;
rtp->lastividtimestamp = 0;
rtp->lastovidtimestamp = 0;
+ rtp->lastitexttimestamp = 0;
+ rtp->lastotexttimestamp = 0;
rtp->lasteventseqn = 0;
rtp->lasteventendseqn = 0;
rtp->lasttxformat = 0;
mark = 1;
}
}
- } else {
+ } else if(f->subclass < AST_FORMAT_MAX_VIDEO) {
mark = f->subclass & 0x1;
pred = rtp->lastovidtimestamp + f->samples;
/* Re-calculate last TS */
rtp->lastovidtimestamp = rtp->lastts;
}
}
+ } else {
+ pred = rtp->lastotexttimestamp + f->samples;
+ /* Re-calculate last TS */
+ rtp->lastts = rtp->lastts + ms * 90;
+ /* If it's close to our prediction, go for it */
+ if (ast_tvzero(f->delivery)) {
+ if (abs(rtp->lastts - pred) < 7200) {
+ rtp->lastts = pred;
+ rtp->lastotexttimestamp += f->samples;
+ } else {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples);
+ rtp->lastotexttimestamp = rtp->lastts;
+ }
+ }
}
/* If the timestamp for non-digit packets has moved beyond the timestamp
for digits, update the digit timestamp.
return 0;
/* Make sure we have enough space for RTP header */
- if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
- ast_log(LOG_WARNING, "RTP can only send voice and video\n");
+ if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO) && (_f->frametype != AST_FRAME_TEXT)) {
+ ast_log(LOG_WARNING, "RTP can only send voice, video and text\n");
return -1;
}
+ /* The bottom bit of a video subclass contains the marker bit */
subclass = _f->subclass;
if (_f->frametype == AST_FRAME_VIDEO)
subclass &= ~0x1;
}
/*! \brief Bridge loop for true native bridge (reinvite) */
-static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, struct ast_rtp_protocol *pr0, struct ast_rtp_protocol *pr1, int codec0, int codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, struct ast_rtp *tp0, struct ast_rtp *tp1, struct ast_rtp_protocol *pr0, struct ast_rtp_protocol *pr1, int codec0, int codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
{
struct ast_frame *fr = NULL;
struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
int oldcodec0 = codec0, oldcodec1 = codec1;
- struct sockaddr_in ac1 = {0,}, vac1 = {0,}, ac0 = {0,}, vac0 = {0,};
- struct sockaddr_in t1 = {0,}, vt1 = {0,}, t0 = {0,}, vt0 = {0,};
+ struct sockaddr_in ac1 = {0,}, vac1 = {0,}, tac1 = {0,}, ac0 = {0,}, vac0 = {0,}, tac0 = {0,};
+ struct sockaddr_in t1 = {0,}, vt1 = {0,}, tt1 = {0,}, t0 = {0,}, vt0 = {0,}, tt0 = {0,};
/* Set it up so audio goes directly between the two endpoints */
/* Test the first channel */
- if (!(pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))) {
+ if (!(pr0->set_rtp_peer(c0, p1, vp1, tp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))) {
ast_rtp_get_peer(p1, &ac1);
if (vp1)
ast_rtp_get_peer(vp1, &vac1);
+ if (tp1)
+ ast_rtp_get_peer(tp1, &tac1);
} else
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
/* Test the second channel */
- if (!(pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))) {
+ if (!(pr1->set_rtp_peer(c1, p0, vp0, tp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))) {
ast_rtp_get_peer(p0, &ac0);
if (vp0)
ast_rtp_get_peer(vp0, &vac0);
+ if (tp0)
+ ast_rtp_get_peer(tp0, &tac0);
} else
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
if (option_debug)
ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
if (c0->tech_pvt == pvt0)
- if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
+ if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
if (c1->tech_pvt == pvt1)
- if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
+ if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
return AST_BRIDGE_RETRY;
}
ast_rtp_get_peer(p1, &t1);
if (vp1)
ast_rtp_get_peer(vp1, &vt1);
+ if (tp1)
+ ast_rtp_get_peer(tp1, &tt1);
if (pr1->get_codec)
codec1 = pr1->get_codec(c1);
ast_rtp_get_peer(p0, &t0);
if (vp0)
ast_rtp_get_peer(vp0, &vt0);
+ if (tp0)
+ ast_rtp_get_peer(tp0, &tt0);
if (pr0->get_codec)
codec0 = pr0->get_codec(c0);
if ((inaddrcmp(&t1, &ac1)) ||
(vp1 && inaddrcmp(&vt1, &vac1)) ||
+ (tp1 && inaddrcmp(&tt1, &tac1)) ||
(codec1 != oldcodec1)) {
if (option_debug > 1) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
+ ast_log(LOG_DEBUG, "Oooh, '%s' changed end taddress to %s:%d (format %d)\n",
+ c1->name, ast_inet_ntoa(tt1.sin_addr), ntohs(tt1.sin_port), codec1);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
+ ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
+ c1->name, ast_inet_ntoa(tac1.sin_addr), ntohs(tac1.sin_port), oldcodec1);
}
- if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
+ if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, tt1.sin_addr.s_addr ? tp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
memcpy(&ac1, &t1, sizeof(ac1));
memcpy(&vac1, &vt1, sizeof(vac1));
+ memcpy(&tac1, &tt1, sizeof(tac1));
oldcodec1 = codec1;
}
if ((inaddrcmp(&t0, &ac0)) ||
- (vp0 && inaddrcmp(&vt0, &vac0))) {
+ (vp0 && inaddrcmp(&vt0, &vac0)) ||
+ (tp0 && inaddrcmp(&tt0, &tac0))) {
if (option_debug > 1) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
}
- if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
+ if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, tt0.sin_addr.s_addr ? tp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
memcpy(&ac0, &t0, sizeof(ac0));
memcpy(&vac0, &vt0, sizeof(vac0));
+ memcpy(&tac0, &tt0, sizeof(tac0));
oldcodec0 = codec0;
}
if (option_debug)
ast_log(LOG_DEBUG, "Oooh, got a %s\n", fr ? "digit" : "hangup");
if (c0->tech_pvt == pvt0)
- if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
+ if (pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
if (c1->tech_pvt == pvt1)
- if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
+ if (pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
return AST_BRIDGE_COMPLETE;
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if (fr->subclass == AST_CONTROL_HOLD) {
/* If we someone went on hold we want the other side to reinvite back to us */
if (who == c0)
- pr1->set_rtp_peer(c1, NULL, NULL, 0, 0);
+ pr1->set_rtp_peer(c1, NULL, NULL, NULL, 0, 0);
else
- pr0->set_rtp_peer(c0, NULL, NULL, 0, 0);
+ pr0->set_rtp_peer(c0, NULL, NULL, NULL, 0, 0);
} else if (fr->subclass == AST_CONTROL_UNHOLD) {
/* If they went off hold they should go back to being direct */
if (who == c0)
- pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE));
+ pr1->set_rtp_peer(c1, p0, vp0, tp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE));
else
- pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE));
+ pr0->set_rtp_peer(c0, p1, vp1, tp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE));
}
ast_indicate_data(other, fr->subclass, fr->data, fr->datalen);
ast_frfree(fr);
{
struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */
+ struct ast_rtp *tp0 = NULL, *tp1 = NULL; /* Text RTP channels */
struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
- enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
- enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
+ enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED, text_p0_res = AST_RTP_GET_FAILED;
+ enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED, text_p1_res = AST_RTP_GET_FAILED;
enum ast_bridge_result res = AST_BRIDGE_FAILED;
int codec0 = 0, codec1 = 0;
void *pvt0 = NULL, *pvt1 = NULL;
/* Get audio and video interface (if native bridge is possible) */
audio_p0_res = pr0->get_rtp_info(c0, &p0);
video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
+ text_p0_res = pr0->get_trtp_info ? pr0->get_trtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
audio_p1_res = pr1->get_rtp_info(c1, &p1);
video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
+ text_p1_res = pr1->get_trtp_info ? pr1->get_trtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
/* If we are carrying video, and both sides are not reinviting... then fail the native bridge */
if (video_p0_res != AST_RTP_GET_FAILED && (audio_p0_res != AST_RTP_TRY_NATIVE || video_p0_res != AST_RTP_TRY_NATIVE))
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
- res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
+ res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, tp0, tp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
}
return res;
projects / thirdparty / asterisk.git / commitdiff
