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-rw-r--r--channels/chan_sip.c451
1 files changed, 396 insertions, 55 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 20a14f0fb..f61ea3a4f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -510,6 +510,7 @@ static const struct cfsip_options {
#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
@@ -563,6 +564,7 @@ static int global_mwitime; /*!< Time between MWI checks for peers */
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 */
@@ -794,10 +796,14 @@ struct sip_auth {
#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 */
@@ -833,6 +839,7 @@ static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_
#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 {
@@ -976,6 +983,7 @@ struct sip_pvt {
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 */
@@ -1010,6 +1018,7 @@ struct sip_pvt {
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 */
@@ -1545,9 +1554,10 @@ static int handle_response_register(struct sip_pvt *p, int resp, char *rest, str
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);
@@ -1571,6 +1581,7 @@ static const struct ast_channel_tech sip_tech = {
.read = sip_read,
.write = sip_write,
.write_video = sip_write,
+ .write_text = sip_write,
.indicate = sip_indicate,
.transfer = sip_transfer,
.fixup = sip_fixup,
@@ -1619,6 +1630,7 @@ static struct ast_rtp_protocol sip_rtp = {
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,
};
@@ -2839,6 +2851,11 @@ static void do_setnat(struct sip_pvt *p, int natflags)
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.
@@ -2860,6 +2877,10 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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;
@@ -2898,6 +2919,13 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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);
@@ -3181,6 +3209,8 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
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)
@@ -3644,10 +3674,13 @@ static int sip_hangup(struct ast_channel *ast)
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);
@@ -3657,11 +3690,15 @@ static int sip_hangup(struct ast_channel *ast)
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. */
@@ -3779,6 +3816,23 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
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;
@@ -4010,7 +4064,10 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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 */
@@ -4040,15 +4097,22 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
/* 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));
@@ -4073,6 +4137,13 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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");
@@ -4096,6 +4167,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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);
}
@@ -4314,6 +4388,19 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
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;
@@ -4331,7 +4418,7 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
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);
}
@@ -4475,11 +4562,15 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
/* 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);
@@ -4502,6 +4593,11 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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;
@@ -4944,6 +5040,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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];
@@ -4952,20 +5049,24 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* 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;
@@ -4991,6 +5092,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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 ? */
@@ -5017,15 +5123,24 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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) ||
@@ -5046,7 +5161,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
} 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;
@@ -5060,6 +5175,22 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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)
@@ -5092,27 +5223,35 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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) {
@@ -5146,11 +5285,15 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
}
}
- /* 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 */
@@ -5240,10 +5383,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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);
+ }
}
}
@@ -5360,21 +5511,23 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* 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),
@@ -5403,6 +5556,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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);
@@ -5431,6 +5586,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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];
@@ -5450,7 +5612,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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);
}
@@ -6187,6 +6349,52 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
*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)
{
@@ -6344,12 +6552,14 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
/*! \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 */
@@ -6383,8 +6593,10 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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 */
@@ -6396,25 +6608,34 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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");
@@ -6434,7 +6655,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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));
}
@@ -6456,8 +6678,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
}
/* 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) {
@@ -6470,8 +6695,36 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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 */
@@ -6529,7 +6782,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
}
/* 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;
@@ -6541,11 +6794,16 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
&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 */
@@ -6565,21 +6823,32 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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);
@@ -6598,6 +6867,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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 ? */
@@ -9359,6 +9633,11 @@ static enum check_auth_result check_user_ok(struct sip_pvt *p, char *of,
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;
@@ -9473,6 +9752,10 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
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;
@@ -9962,6 +10245,7 @@ static int _sip_show_peers(int fd, int *total, struct mansession *s, const struc
"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",
@@ -9972,6 +10256,7 @@ static int _sip_show_peers(int fd, int *total, struct mansession *s, const struc
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");
@@ -10392,6 +10677,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, const struct m
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");
@@ -10479,6 +10765,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, const struct m
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)));
@@ -10628,6 +10915,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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");
@@ -10652,6 +10940,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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");
@@ -11262,6 +11551,7 @@ static int sip_do_debug(int fd, int argc, char *argv[])
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;
@@ -11329,6 +11619,7 @@ static int sip_no_debug(int fd, int argc, char *argv[])
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;
}
@@ -12510,6 +12801,8 @@ static void stop_media_flows(struct sip_pvt *p)
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);
}
@@ -14487,7 +14780,7 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
/* 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))
@@ -14502,6 +14795,13 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
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 */
@@ -15697,6 +15997,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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);
@@ -15819,6 +16120,10 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
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);
@@ -16517,6 +16822,7 @@ static int reload_config(enum channelreloadreason reason)
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;
@@ -16584,6 +16890,7 @@ static int reload_config(enum channelreloadreason reason)
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) {
@@ -16806,6 +17113,9 @@ static int reload_config(enum channelreloadreason reason)
} 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);
@@ -17208,8 +17518,33 @@ static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struc
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;
@@ -17249,6 +17584,12 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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;
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-04 07:04:36 +0000