[asterisk-commits] mnicholson: branch 1.4 r227758 - /branches/1.4/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Nov 4 13:55:49 CST 2009
Author: mnicholson
Date: Wed Nov 4 13:55:44 2009
New Revision: 227758
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=227758
Log:
Modify the SDP parsing code to parse session and media level items separately.
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future.
(closes issue #14994)
Reported by: frawd
Tested by: frawd, mnicholson, file
Review: https://reviewboard.asterisk.org/r/385/
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=227758&r1=227757&r2=227758
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Nov 4 13:55:44 2009
@@ -1353,10 +1353,15 @@
/*--- Codec handling / SDP */
static void try_suggested_sip_codec(struct sip_pvt *p);
-static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
-static const char *get_sdp(struct sip_request *req, const char *name);
+static const char *get_sdp_iterate(int* start, struct sip_request *req, const char *name);
+static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req);
+static int process_sdp_c(const char *c, struct ast_hostent *hp);
+static int process_sdp_a_sendonly(const char *a, int *sendonly);
+static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp *newaudiortp, int *last_rtpmap_codec);
+static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp *newvideortp, int *last_rtpmap_codec);
+static int process_sdp_a_image(const char *a, struct sip_pvt *p);
static void add_codec_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);
@@ -4383,12 +4388,27 @@
return "";
}
-/*! \brief Get a line from an SDP message body */
-static const char *get_sdp(struct sip_request *req, const char *name)
-{
- int dummy = 0;
-
- return get_sdp_iterate(&dummy, req, name);
+/*! \brief Fetches the next valid SDP line between the 'start' line
+ * and the 'stop' line. Returns the type ('a', 'c', ...) and
+ * matching line in reference 'start' is updated with the next line number.
+ */
+static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value)
+{
+ char type = '\0';
+ const char *line = NULL;
+
+ if (stop > req->sdp_end || stop < req->sdp_start) stop = req->sdp_end;
+
+ while (*start < stop) {
+ line = req->line[(*start)++];
+ if (line[1] == '=') {
+ type = line[0];
+ *value = ast_skip_blanks(line + 2);
+ break;
+ }
+ }
+
+ return type;
}
/*! \brief Get a specific line from the message body */
@@ -5298,44 +5318,50 @@
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req)
{
- const char *m; /* SDP media offer */
- const char *c;
- const char *a;
- char host[258];
+ /* Iterators for SDP parsing */
+ int start = req->sdp_start;
+ int next = start;
+ int iterator = start;
+
+ /* Temporary vars for SDP parsing */
+ char type = '\0';
+ const char *value = NULL;
+ const char *m = NULL; /* SDP media offer */
+ const char *nextm = NULL;
int len = -1;
+
+ /* Host information */
+ struct ast_hostent audiohp;
+ struct ast_hostent videohp;
+ struct ast_hostent sessionhp;
+ struct hostent *hp = NULL; /*!< RTP Audio host IP */
+ struct hostent *vhp = NULL; /*!< RTP video host IP */
int portno = -1; /*!< RTP Audio port number */
int vportno = -1; /*!< RTP Video port number */
- int udptlportno = -1;
- int peert38capability = 0;
- char s[256];
- int old = 0;
+ int udptlportno = -1; /*!< UDPTL Image port number */
+ struct sockaddr_in sin; /*!< media socket address */
+ struct sockaddr_in vsin; /*!< video socket address */
+ struct sockaddr_in isin; /*!< image socket address */
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
int peercapability = 0, peernoncodeccapability = 0;
int vpeercapability = 0, vpeernoncodeccapability = 0;
- struct sockaddr_in sin; /*!< media socket address */
- struct sockaddr_in vsin; /*!< Video socket address */
-
- const char *codecs;
- struct hostent *hp; /*!< RTP Audio host IP */
- struct hostent *vhp = NULL; /*!< RTP video host IP */
- struct ast_hostent audiohp;
- struct ast_hostent videohp;
- int codec;
- int destiterator = 0;
- int iterator;
- int sendonly = -1;
- int numberofports;
- struct ast_rtp *newaudiortp, *newvideortp; /* Buffers for codec handling */
- int newjointcapability; /* Negotiated capability */
+ struct ast_rtp *newaudiortp, *newvideortp; /*!< Buffers for codec handling */
+ int newjointcapability; /*!< Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
+ const char *codecs;
+ int codec;
+
+ /* Others */
+ int sendonly = -1;
+ int vsendonly = -1;
+ int numberofports;
int numberofmediastreams = 0;
+ int last_rtpmap_codec = 0;
int debug = sip_debug_test_pvt(p);
-
- int found_rtpmap_codecs[SDP_MAX_RTPMAP_CODECS];
- int last_rtpmap_codec=0;
-
+
+ /* Initial check */
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1;
@@ -5360,52 +5386,75 @@
ast_rtp_new_init(newvideortp);
ast_rtp_pt_clear(newvideortp);
+
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
+ ast_set_flag(&p->flags[0], SIP_NOVIDEO);
+
memset(p->offered_media, 0, sizeof(p->offered_media));
- /* Try to find first media stream */
- m = get_sdp(req, "m");
- destiterator = req->sdp_start;
- c = get_sdp_iterate(&destiterator, req, "c");
- if (ast_strlen_zero(m) || ast_strlen_zero(c)) {
- ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c);
+
+ /* Scan for the first media stream (m=) line to limit scanning of globals */
+ nextm = get_sdp_iterate(&next, req, "m");
+ if (ast_strlen_zero(nextm)) {
+ ast_log(LOG_WARNING, "Insufficient information for SDP (m= not found)\n");
return -1;
}
- /* Check for IPv4 address (not IPv6 yet) */
- if (sscanf(c, "IN IP4 %256s", host) != 1) {
- ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
- return -1;
- }
-
- /* XXX This could block for a long time, and block the main thread! XXX */
- hp = ast_gethostbyname(host, &audiohp);
- if (!hp) {
- ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
- return -1;
- }
- vhp = hp; /* Copy to video address as default too */
-
- iterator = req->sdp_start;
- ast_set_flag(&p->flags[0], SIP_NOVIDEO);
-
-
- /* Find media streams in this SDP offer */
- while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
+ /* Scan session level SDP parameters (lines before first media stream) */
+ while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
+ int processed = FALSE;
+ switch (type) {
+ case 'c':
+ if (process_sdp_c(value, &sessionhp)) {
+ processed = TRUE;
+ hp = &sessionhp.hp;
+ vhp = hp;
+ }
+ break;
+ case 'a':
+ if (process_sdp_a_sendonly(value, &sendonly)) {
+ processed = TRUE;
+ vsendonly = sendonly;
+ }
+ else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec))
+ processed = TRUE;
+ else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec))
+ processed = TRUE;
+ else if (process_sdp_a_image(value, p))
+ processed = TRUE;
+ break;
+ }
+
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED.");
+ }
+
+
+ /* Scan media stream (m=) specific parameters loop */
+ while (!ast_strlen_zero(nextm)) {
+ int audio = FALSE;
+ int video = FALSE;
+ int image = FALSE;
int x;
- int audio = FALSE;
numberofports = 1;
len = -1;
+ start = next;
+ m = nextm;
+ iterator = next;
+ nextm = get_sdp_iterate(&next, req, "m");
+
+ /* Search for audio media definition */
if ((sscanf(m, "audio %30d/%30d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
(sscanf(m, "audio %30d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
+ /* Found audio stream in this media definition */
audio = TRUE;
p->offered_media[SDP_AUDIO].offered = TRUE;
numberofmediastreams++;
- /* Found audio stream in this media definition */
portno = x;
+
/* Scan through the RTP payload types specified in a "m=" line: */
codecs = m + len;
ast_copy_string(p->offered_media[SDP_AUDIO].text, codecs, sizeof(p->offered_media[SDP_AUDIO].text));
@@ -5418,13 +5467,16 @@
ast_verbose("Found RTP audio format %d\n", codec);
ast_rtp_set_m_type(newaudiortp, codec);
}
+ /* Search for video media definition */
} else if ((sscanf(m, "video %30d/%30d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
- (sscanf(m, "video %30d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
- /* If it is not audio - is it video ? */
+ (sscanf(m, "video %30d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
+ /* Found video stream in this media definition */
+ video = TRUE;
ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
p->offered_media[SDP_VIDEO].offered = TRUE;
numberofmediastreams++;
vportno = x;
+
/* Scan through the RTP payload types specified in a "m=" line: */
codecs = m + len;
ast_copy_string(p->offered_media[SDP_VIDEO].text, codecs, sizeof(p->offered_media[SDP_VIDEO].text));
@@ -5437,8 +5489,11 @@
ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_set_m_type(newvideortp, codec);
}
+ /* Search for image media definition */
} else if (p->udptl && ((sscanf(m, "image %30d udptl t38%n", &x, &len) == 1 && len > 0) ||
(sscanf(m, "image %30d UDPTL t38%n", &x, &len) == 1 && len >= 0))) {
+ /* Found image stream in this media definition */
+ image = TRUE;
if (debug)
ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
p->offered_media[SDP_IMAGE].offered = TRUE;
@@ -5446,7 +5501,7 @@
numberofmediastreams++;
if (p->owner && p->lastinvite) {
- if(p->t38.state != T38_LOCAL_REINVITE) {
+ if (p->t38.state != T38_LOCAL_REINVITE) {
p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */
if (option_debug > 1)
ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" );
@@ -5457,32 +5512,73 @@
if (option_debug > 1)
ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
}
- } else
+ } else {
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
+ continue;
+ }
+
+ /* Check for number of ports */
if (numberofports > 1)
ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
- /* Check for Media-description-level-address for audio */
- c = get_sdp_iterate(&destiterator, req, "c");
- if (!ast_strlen_zero(c)) {
- if (sscanf(c, "IN IP4 %256s", host) != 1) {
- ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
- } else {
- /* XXX This could block for a long time, and block the main thread! XXX */
+
+ /* Media stream specific parameters */
+ while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
+ int processed = FALSE;
+
+ switch (type) {
+ case 'c':
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);
- return -2;
+ if (process_sdp_c(value, &audiohp)) {
+ processed = TRUE;
+ hp = &audiohp.hp;
}
- } else if (!(vhp = ast_gethostbyname(host, &videohp))) {
- ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
- return -2;
}
+ else if (video) {
+ if (process_sdp_c(value, &videohp)) {
+ processed = TRUE;
+ vhp = &videohp.hp;
+ }
+ }
+ break;
+ case 'a':
+ /* Audio specific scanning */
+ if (audio) {
+ if (process_sdp_a_sendonly(value, &sendonly))
+ processed = TRUE;
+ else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec))
+ processed = TRUE;
+ }
+ /* Video specific scanning */
+ else if (video) {
+ if (process_sdp_a_sendonly(value, &vsendonly))
+ processed = TRUE;
+ else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec))
+ processed = TRUE;
+ }
+ /* Image (T.38 FAX) specific scanning */
+ else if (image) {
+ if (process_sdp_a_image(value, p))
+ processed = TRUE;
+ }
+ break;
}
- }
- }
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Processing media-level (%s) SDP %c=%s... %s\n",
+ (audio == TRUE)? "audio" : (video == TRUE)? "video" : "image",
+ type, value,
+ (processed == TRUE)? "OK." : "UNSUPPORTED.");
+ }
+ }
+
+ /* Sanity checks */
+ if (!hp) {
+ ast_log(LOG_WARNING, "Insufficient information in SDP (c=)...\n");
+ return -1;
+ }
+
if (portno == -1 && vportno == -1 && udptlportno == -1)
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
@@ -5492,309 +5588,13 @@
/* We have too many fax, audio and/or video 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;
- memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
- if (vhp)
- memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
-
- /* Setup UDPTL port number */
- if (p->udptl) {
- if (udptlportno > 0) {
- sin.sin_port = htons(udptlportno);
- if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
- struct sockaddr_in peer;
- ast_rtp_get_peer(p->rtp, &peer);
- if (peer.sin_addr.s_addr) {
- memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(sin.sin_addr));
- if (debug) {
- ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
- }
- }
- }
- ast_udptl_set_peer(p->udptl, &sin);
- if (debug)
- ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
- } else {
- ast_udptl_stop(p->udptl);
- if (debug)
- ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n");
- }
- }
-
-
- if (p->rtp) {
- if (portno > 0) {
- sin.sin_port = htons(portno);
- ast_rtp_set_peer(p->rtp, &sin);
- if (debug)
- ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
- } else {
- if (udptlportno > 0) {
- if (debug)
- ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid);
- } else {
- ast_rtp_stop(p->rtp);
- if (debug)
- ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
- }
- }
- }
- /* Setup video port number */
- if (vportno != -1)
- vsin.sin_port = htons(vportno);
-
- /* Next, scan through each "a=rtpmap:" 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 */
- iterator = req->sdp_start;
- while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
- char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
- if (option_debug > 1) {
- int breakout = FALSE;
-
- /* If we're debugging, check for unsupported sdp options */
- if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
- if (debug)
- ast_verbose("Got unsupported a:rtcp in SDP offer \n");
- breakout = TRUE;
- } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
- /* Format parameters: Not supported */
- /* Note: This is used for codec parameters, like bitrate for
- G722 and video formats for H263 and H264
- See RFC2327 for an example */
- if (debug)
- ast_verbose("Got unsupported a:fmtp in SDP offer \n");
- breakout = TRUE;
- } else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
- /* Video stuff: Not supported */
- if (debug)
- ast_verbose("Got unsupported a:framerate in SDP offer \n");
- breakout = TRUE;
- } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
- /* Video stuff: Not supported */
- if (debug)
- ast_verbose("Got unsupported a:maxprate in SDP offer \n");
- breakout = TRUE;
- } else if (!strncasecmp(a, "crypto:", (size_t) 7)) {
- /* SRTP stuff, not yet supported */
- if (debug)
- ast_verbose("Got unsupported a:crypto in SDP offer \n");
- breakout = TRUE;
- }
- if (breakout) /* We have a match, skip to next header */
- continue;
- }
- if (!strcasecmp(a, "sendonly")) {
- if (sendonly == -1)
- sendonly = 1;
- continue;
- } else if (!strcasecmp(a, "inactive")) {
- if (sendonly == -1)
- sendonly = 2;
- continue;
- } else if (!strcasecmp(a, "sendrecv")) {
- if (sendonly == -1)
- sendonly = 0;
- continue;
- } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
- char *tmp = strrchr(a, ':');
- long int framing = 0;
- if (tmp) {
- tmp++;
- framing = strtol(tmp, NULL, 10);
- if (framing == LONG_MIN || framing == LONG_MAX) {
- framing = 0;
- if (option_debug)
- ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
- }
- }
- if (framing && p->autoframing) {
- struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
- int codec_n;
- int format = 0;
- for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) {
- format = ast_rtp_codec_getformat(codec_n);
- if (!format) /* non-codec or not found */
- continue;
- if (option_debug)
- ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
- ast_codec_pref_setsize(pref, format, framing);
- }
- ast_rtp_codec_setpref(p->rtp, pref);
- }
- continue;
- } else if (sscanf(a, "rtpmap: %30u %[^/]/", &codec, mimeSubtype) == 2) {
- /* We have a rtpmap to handle */
- int found = FALSE;
- /* We should propably check if this is an audio or video codec
- so we know where to look */
-
- if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- /* Note: should really look at the 'freq' and '#chans' params too */
- if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
- if (debug)
- ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
- found_rtpmap_codecs[last_rtpmap_codec] = codec;
- last_rtpmap_codec++;
- found = TRUE;
-
- } else if (p->vrtp) {
- if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
- if (debug)
- ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
- found_rtpmap_codecs[last_rtpmap_codec] = codec;
- last_rtpmap_codec++;
- found = TRUE;
- }
- }
- } else {
- if (debug)
- ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
- }
-
- if (!found) {
- /* Remove this codec since it's an unknown media type for us */
- /* XXX This is buggy since the media line for audio and video can have the
- same numbers. We need to check as described above, but for testing this works... */
- ast_rtp_unset_m_type(newaudiortp, codec);
- ast_rtp_unset_m_type(newvideortp, codec);
- if (debug)
- ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
- }
- }
- }
-
- if (udptlportno != -1) {
- int found = 0, x;
-
- old = 0;
-
- /* Scan trough the a= lines for T38 attributes and set apropriate fileds */
- iterator = req->sdp_start;
- while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
- if ((sscanf(a, "T38FaxMaxBuffer:%30d", &x) == 1)) {
- found = 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "MaxBufferSize:%d\n",x);
- } else if ((sscanf(a, "T38MaxBitRate:%30d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%30d", &x) == 1)) {
- found = 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x);
- switch (x) {
- case 14400:
- peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
- break;
- case 12000:
- peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
- break;
- case 9600:
- peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
- break;
- case 7200:
- peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
- break;
- case 4800:
- peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400;
- break;
- case 2400:
- peert38capability |= T38FAX_RATE_2400;
- break;
- }
- } else if ((sscanf(a, "T38FaxVersion:%30d", &x) == 1)) {
- found = 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "FaxVersion: %d\n",x);
- if (x == 0)
- peert38capability |= T38FAX_VERSION_0;
- else if (x == 1)
- peert38capability |= T38FAX_VERSION_1;
- } else if ((sscanf(a, "T38FaxMaxDatagram:%30d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%30d", &x) == 1)) {
- found = 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "FaxMaxDatagram: %d\n",x);
- ast_udptl_set_far_max_datagram(p->udptl, x);
- ast_udptl_set_local_max_datagram(p->udptl, x);
- } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
- found = 1;
- if ((sscanf(a, "T38FaxFillBitRemoval:%30d", &x) == 1)) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "FillBitRemoval: %d\n",x);
- if (x == 1)
- peert38capability |= T38FAX_FILL_BIT_REMOVAL;
- } else {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "FillBitRemoval\n");
- peert38capability |= T38FAX_FILL_BIT_REMOVAL;
- }
- } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
- found = 1;
- if ((sscanf(a, "T38FaxTranscodingMMR:%30d", &x) == 1)) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Transcoding MMR: %d\n",x);
- if (x == 1)
- peert38capability |= T38FAX_TRANSCODING_MMR;
- } else {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Transcoding MMR\n");
- peert38capability |= T38FAX_TRANSCODING_MMR;
- }
- } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
- found = 1;
- if ((sscanf(a, "T38FaxTranscodingJBIG:%30d", &x) == 1)) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Transcoding JBIG: %d\n",x);
- if (x == 1)
- peert38capability |= T38FAX_TRANSCODING_JBIG;
- } else {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Transcoding JBIG\n");
- peert38capability |= T38FAX_TRANSCODING_JBIG;
- }
- } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
- found = 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "RateManagement: %s\n", s);
- if (!strcasecmp(s, "localTCF"))
- peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF;
- else if (!strcasecmp(s, "transferredTCF"))
- peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
- } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
- found = 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "UDP EC: %s\n", s);
- if (!strcasecmp(s, "t38UDPRedundancy")) {
- peert38capability |= T38FAX_UDP_EC_REDUNDANCY;
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
- } else if (!strcasecmp(s, "t38UDPFEC")) {
- peert38capability |= T38FAX_UDP_EC_FEC;
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
- } else {
- peert38capability |= T38FAX_UDP_EC_NONE;
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
- }
- }
- }
- if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */
- p->t38.peercapability = peert38capability;
- p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */
- peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400);
- p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */
- }
- if (debug)
- ast_log(LOG_DEBUG, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n",
- p->t38.capability,
- p->t38.peercapability,
- p->t38.jointcapability);
- } else {
+ if (udptlportno == -1) {
p->t38.state = T38_DISABLED;
if (option_debug > 2)
ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
}
+
/* 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);
@@ -5834,7 +5634,7 @@
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
- p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
+ p->jointcapability = newjointcapability; /* Our joint codec profile */
p->peercapability = newpeercapability; /* The other sides capability in latest offer */
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
@@ -5856,18 +5656,65 @@
}
/* Setup audio port number */
- if (p->rtp && sin.sin_port) {
- ast_rtp_set_peer(p->rtp, &sin);
- if (debug)
- ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+ if (p->rtp) {
+ if (portno > 0) {
+ sin.sin_family = AF_INET;
+ sin.sin_port = htons(portno);
+ memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
+ ast_rtp_set_peer(p->rtp, &sin);
+ if (debug)
+ ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+ } else if (udptlportno > 0) {
+ if (debug)
+ ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
+ } else {
+ ast_rtp_stop(p->rtp);
+ if (debug)
+ ast_verbose("Peer doesn't provide audio\n");
+ }
}
/* Setup video port number */
- if (p->vrtp && vsin.sin_port) {
- ast_rtp_set_peer(p->vrtp, &vsin);
- if (debug)
- ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
- }
+ if (p->vrtp) {
+ if (vportno > 0) {
+ vsin.sin_family = AF_INET;
+ vsin.sin_port = htons(vportno);
+ memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
+ ast_rtp_set_peer(p->vrtp, &vsin);
+ if (debug)
+ ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
+ } else {
+ ast_rtp_stop(p->vrtp);
+ if (debug)
+ ast_verbose("Peer doesn't provide video\n");
+ }
+ }
+
+ /* Setup image port number */
+ if (p->udptl) {
+ if (udptlportno > 0) {
+ isin.sin_family = AF_INET;
+ isin.sin_port = htons(udptlportno);
+ if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
+ struct sockaddr_in peer;
+ ast_rtp_get_peer(p->rtp, &peer);
+ if (peer.sin_addr.s_addr) {
+ memcpy(&isin.sin_addr, &peer.sin_addr, sizeof(isin.sin_addr));
+ if (debug)
+ ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(isin.sin_addr));
+ }
+ } else
+ memcpy(&isin.sin_addr, hp->h_addr, sizeof(isin.sin_addr));
+ ast_udptl_set_peer(p->udptl, &isin);
+ if (debug)
+ ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(isin.sin_addr), ntohs(isin.sin_port));
+ } else {
+ ast_udptl_stop(p->udptl);
+ if (debug)
+ ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n");
+ }
+ }
+
/* Ok, we're going with this offer */
if (option_debug > 1) {
@@ -5892,7 +5739,8 @@
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
-
+
+ /* sendonly processing */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && sin.sin_addr.s_addr && (!sendonly || sendonly == -1)) {
ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
/* Activate a re-invite */
@@ -5913,8 +5761,271 @@
change_hold_state(p, req, FALSE, sendonly);
else if (!sin.sin_addr.s_addr || (sendonly && sendonly != -1))
change_hold_state(p, req, TRUE, sendonly);
+
return 0;
}
+
+
+static int process_sdp_c(const char *c, struct ast_hostent *ast_hp)
+{
+ char host[258];
+ struct hostent *hp;
+
+ /* Check for Media-description-level-address */
+ if (sscanf(c, "IN IP4 %255s", host) != 1) {
+ ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
+ return FALSE;
+ } else {
+ if (!(hp = ast_gethostbyname(host, ast_hp))) {
+ ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in c= line, '%s'\n", c);
+ return FALSE;
+ }
+ return TRUE;
+ }
+ return FALSE;
+}
+
+static int process_sdp_a_sendonly(const char *a, int *sendonly)
+{
+ int found = FALSE;
+
+ if (!strcasecmp(a, "sendonly")) {
+ if (*sendonly == -1)
+ *sendonly = 1;
+ found = TRUE;
+ } else if (!strcasecmp(a, "inactive")) {
+ if (*sendonly == -1)
+ *sendonly = 2;
+ found = TRUE;
+ } else if (!strcasecmp(a, "sendrecv")) {
+ if (*sendonly == -1)
+ *sendonly = 0;
+ found = TRUE;
+ }
+ return found;
+}
+
+static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp *newaudiortp, int *last_rtpmap_codec)
+{
+ int found = FALSE;
+ int codec;
+ char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
+ int debug = sip_debug_test_pvt(p);
+
+ if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
+ char *tmp = strrchr(a, ':');
+ long int framing = 0;
+ if (tmp) {
+ tmp++;
+ framing = strtol(tmp, NULL, 10);
+ if (framing == LONG_MIN || framing == LONG_MAX) {
+ framing = 0;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
+ }
+ }
+ if (framing && p->autoframing) {
+ struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ int codec_n;
+ int format = 0;
+ for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) {
+ format = ast_rtp_codec_getformat(codec_n);
+ if (!format) /* non-codec or not found */
+ continue;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
+ ast_codec_pref_setsize(pref, format, framing);
+ }
+ ast_rtp_codec_setpref(p->rtp, pref);
+ }
+ found = TRUE;
+ } else if (sscanf(a, "rtpmap: %30u %[^/]/", &codec, mimeSubtype) == 2) {
+ /* We have a rtpmap to handle */
+ if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
+ /* Note: should really look at the 'freq' and '#chans' params too */
+ if (ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
+ if (debug)
+ ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
+ (*last_rtpmap_codec)++;
+ found = TRUE;
+ }
+ } else {
+ if (debug)
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
+ }
+
+ if (!found) {
+ /* Remove this codec since it's an unknown media type for us */
+ ast_rtp_unset_m_type(newaudiortp, codec);
+ if (debug)
+ ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ }
+
+ return found;
+}
+
+static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp *newvideortp, int *last_rtpmap_codec)
+{
+ int found = FALSE;
+ int codec;
+ char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
+ int debug = sip_debug_test_pvt(p);
+
+ if (sscanf(a, "rtpmap: %30u %[^/]/", &codec, mimeSubtype) == 2) {
+ /* We have a rtpmap to handle */
+ if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
+ /* Note: should really look at the 'freq' and '#chans' params too */
+ if (p->vrtp && ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
+ if (debug)
+ ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
+ (*last_rtpmap_codec)++;
+ found = TRUE;
+ }
+ } else {
+ if (debug)
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
+ }
+
+ if (!found) {
+ /* Remove this codec since it's an unknown media type for us */
+ ast_rtp_unset_m_type(newvideortp, codec);
+ if (debug)
+ ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ }
+
+ return found;
+}
+
+static int process_sdp_a_image(const char *a, struct sip_pvt *p)
+{
+ int found = FALSE;
+ int peert38capability = 0;
+ char s[256];
+ int x;
+ int debug = sip_debug_test_pvt(p);
+
+ /* Scan trough the a= lines for T38 attributes and set apropriate fileds */
+ if ((sscanf(a, "T38FaxMaxBuffer:%30d", &x) == 1)) {
+ found = TRUE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "MaxBufferSize:%d\n",x);
+ } else if ((sscanf(a, "T38MaxBitRate:%30d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%30d", &x) == 1)) {
+ found = TRUE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x);
+ switch (x) {
+ case 14400:
+ peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 12000:
+ peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 9600:
+ peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 7200:
+ peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 4800:
+ peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400;
+ break;
+ case 2400:
+ peert38capability |= T38FAX_RATE_2400;
+ break;
+ }
+ } else if ((sscanf(a, "T38FaxVersion:%30d", &x) == 1)) {
+ found = TRUE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "FaxVersion: %d\n",x);
+ if (x == 0)
+ peert38capability |= T38FAX_VERSION_0;
+ else if (x == 1)
+ peert38capability |= T38FAX_VERSION_1;
+ } else if ((sscanf(a, "T38FaxMaxDatagram:%30d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%30d", &x) == 1)) {
+ found = TRUE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "FaxMaxDatagram: %d\n",x);
+ ast_udptl_set_far_max_datagram(p->udptl, x);
+ ast_udptl_set_local_max_datagram(p->udptl, x);
+ } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
+ found = TRUE;
+ if ((sscanf(a, "T38FaxFillBitRemoval:%30d", &x) == 1)) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "FillBitRemoval: %d\n",x);
+ if (x == 1)
+ peert38capability |= T38FAX_FILL_BIT_REMOVAL;
+ } else {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "FillBitRemoval\n");
+ peert38capability |= T38FAX_FILL_BIT_REMOVAL;
+ }
+ } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
+ found = TRUE;
+ if ((sscanf(a, "T38FaxTranscodingMMR:%30d", &x) == 1)) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Transcoding MMR: %d\n",x);
+ if (x == 1)
+ peert38capability |= T38FAX_TRANSCODING_MMR;
+ } else {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Transcoding MMR\n");
+ peert38capability |= T38FAX_TRANSCODING_MMR;
+ }
+ } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
+ found = TRUE;
+ if ((sscanf(a, "T38FaxTranscodingJBIG:%30d", &x) == 1)) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Transcoding JBIG: %d\n",x);
+ if (x == 1)
+ peert38capability |= T38FAX_TRANSCODING_JBIG;
+ } else {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Transcoding JBIG\n");
+ peert38capability |= T38FAX_TRANSCODING_JBIG;
+ }
+ } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
+ found = TRUE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "RateManagement: %s\n", s);
+ if (!strcasecmp(s, "localTCF"))
+ peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF;
+ else if (!strcasecmp(s, "transferredTCF"))
+ peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
+ } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
+ found = TRUE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "UDP EC: %s\n", s);
+ if (!strcasecmp(s, "t38UDPRedundancy")) {
+ peert38capability |= T38FAX_UDP_EC_REDUNDANCY;
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
+ } else if (!strcasecmp(s, "t38UDPFEC")) {
+ peert38capability |= T38FAX_UDP_EC_FEC;
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
+ } else {
+ peert38capability |= T38FAX_UDP_EC_NONE;
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
+ }
+ }
+
+ if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */
+ p->t38.peercapability = peert38capability;
+ p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */
+ peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400);
+ p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */
+ }
+ if (debug)
+ ast_log(LOG_DEBUG, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n",
+ p->t38.capability,
+ p->t38.peercapability,
+ p->t38.jointcapability);
+
+ return found;
+}
+
+
#ifdef LOW_MEMORY
static void ts_ast_rtp_destroy(void *data)
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