diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d129d1e718c3c13976d07a4ae0573c7acdcb8686..a7141b6e1e5dd5c8a75be5b25324dc871d314e11 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2497,10 +2497,17 @@ static int is_method_allowed(unsigned int *allowed_methods, enum sipmethod metho
/*--- 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, int t38action);
+static int process_sdp_o(const char *o, struct sip_pvt *p);
+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_codecs *newaudiortp, int *last_rtpmap_codec);
+static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
+static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, 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, format_t codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size);
@@ -7128,12 +7135,27 @@ static const char *get_sdp_iterate(int *start, struct sip_request *req, const ch
return "";
}
-/*! \brief Get a line from an SDP message body */
-static const char *get_sdp(struct sip_request *req, const char *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)
{
- int dummy = 0;
+ 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_OFFSET_TO_STR(req, line[(*start)++]);
+ if (line[1] == '=') {
+ type = line[0];
+ *value = ast_skip_blanks(line + 2);
+ break;
+ }
+ }
- return get_sdp_iterate(&dummy, req, name);
+ return type;
}
/*! \brief Get a specific line from the message body */
@@ -8348,60 +8370,63 @@ static int get_ip_and_port_from_sdp(struct sip_request *req, const enum media_ty
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action)
{
- const char *m; /* SDP media offer */
- const char *c;
- const char *a;
- const char *o; /* Pointer to o= line */
- char *o_copy; /* Copy of o= line */
- char *token;
- 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;
- int portno = -1; /*!< RTP Audio port number */
- int vportno = -1; /*!< RTP Video port number */
+
+ /* Host information */
+ struct ast_hostent sessionhp;
+ struct ast_hostent audiohp;
+ struct ast_hostent videohp;
+ struct ast_hostent texthp;
+ struct hostent *hp = NULL; /*!< RTP Audio host IP */
+ struct hostent *vhp = NULL; /*!< RTP video host IP */
+ struct hostent *thp = NULL; /*!< RTP text host IP */
+ int portno = -1; /*!< RTP Audio port number */
+ int vportno = -1; /*!< RTP Video port number */
int tportno = -1; /*!< RTP Text port number */
- int udptlportno = -1;
- 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 */
+ struct sockaddr_in tsin; /*!< text socket address */
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
format_t peercapability = 0, vpeercapability = 0, tpeercapability = 0;
int peernoncodeccapability = 0, vpeernoncodeccapability = 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 = -1;
- int numberofports;
struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
format_t newjointcapability; /* Negotiated capability */
format_t 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 red_data_pt[10]; /* For T.140 red */
+ int red_num_gen = 0; /* For T.140 red */
+ char red_fmtp[100] = "empty"; /* For T.140 red */
int debug = sip_debug_test_pvt(p);
-
- int found_rtpmap_codecs[SDP_MAX_RTPMAP_CODECS];
- int last_rtpmap_codec=0;
+ /* START UNKNOWN */
char buf[SIPBUFSIZE];
- int64_t rua_version;
-
- int red_data_pt[10];
- int red_num_gen = 0;
- int red_pt = 0;
-
- char *red_cp; /* For T.140 red */
- char red_fmtp[100] = "empty"; /* For T.140 red */
+ /* END UNKNOWN */
+ /* Initial check */
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1;
@@ -8417,102 +8442,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
memset(p->offered_media, 0, sizeof(p->offered_media));
- /* Store the SDP version number of remote UA. This will allow us to
- distinguish between session modifications and session refreshes. If
- the remote UA does not send an incremented SDP version number in a
- subsequent RE-INVITE then that means its not changing media session.
- The RE-INVITE may have been sent to update connected party, remote
- target or to refresh the session (Session-Timers). Asterisk must not
- change media session and increment its own version number in answer
- SDP in this case. */
-
- o = get_sdp(req, "o");
- if (ast_strlen_zero(o)) {
- ast_log(LOG_WARNING, "SDP syntax error. SDP without an o= line\n");
- return -1;
- }
-
- o_copy = ast_strdupa(o);
- token = strsep(&o_copy, " "); /* Skip username */
- if (!o_copy) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line username\n");
- return -1;
- }
- token = strsep(&o_copy, " "); /* Skip session-id */
- if (!o_copy) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line session-id\n");
- return -1;
- }
- token = strsep(&o_copy, " "); /* Version */
- if (!o_copy) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line\n");
- return -1;
- }
- if (!sscanf(token, "%30" SCNd64, &rua_version)) {
- ast_log(LOG_WARNING, "SDP syntax error in o= line version\n");
- return -1;
- }
-
- /* we need to check the SDP version number the other end sent us;
- * our rules for deciding what to accept are a bit complex.
- *
- * 1) if 'ignoresdpversion' has been set for this dialog, then
- * we will just accept whatever they sent and assume it is
- * a modification of the session, even if it is not
- * 2) otherwise, if this is the first SDP we've seen from them
- * we accept it
- * 3) otherwise, if the new SDP version number is higher than the
- * old one, we accept it
- * 4) otherwise, if this SDP is in response to us requesting a switch
- * to T.38, we accept the SDP, but also generate a warning message
- * that this peer should have the 'ignoresdpversion' option set,
- * because it is not following the SDP offer/answer RFC; if we did
- * not request a switch to T.38, then we stop parsing the SDP, as it
- * has not changed from the previous version
- */
-
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION) ||
- (p->sessionversion_remote < 0) ||
- (p->sessionversion_remote < rua_version)) {
- p->sessionversion_remote = rua_version;
- p->session_modify = TRUE;
- } else {
- if (p->t38.state == T38_LOCAL_REINVITE) {
- p->sessionversion_remote = rua_version;
- p->session_modify = TRUE;
- ast_log(LOG_WARNING, "Call %s responded to our T.38 reinvite without changing SDP version; 'ignoresdpversion' should be set for this peer.\n", p->callid);
- } else {
- p->session_modify = FALSE;
- ast_debug(2, "Call %s responded to our reinvite without changing SDP version; ignoring SDP.\n", p->callid);
- return 0;
- }
- }
-
- /* 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);
- 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 */
- thp = hp; /* Copy to text address as default too */
-
- iterator = req->sdp_start;
/* default: novideo and notext set */
p->novideo = TRUE;
p->notext = TRUE;
@@ -8525,22 +8455,74 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
}
- /* Find media streams in this SDP offer */
- while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
- int x;
+ /* 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;
+ }
+
+ /* 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 'o':
+ if (!process_sdp_o(value, p))
+ return -1;
+ break;
+ case 'c':
+ if (process_sdp_c(value, &sessionhp)) {
+ processed = TRUE;
+ hp = &sessionhp.hp;
+ vhp = hp;
+ thp = 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_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &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 text = FALSE;
+ int x;
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)) {
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].codecs, codecs, sizeof(p->offered_media[SDP_AUDIO].codecs));
@@ -8554,13 +8536,15 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, 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)) {
+ (sscanf(m, "video %30d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
video = TRUE;
p->novideo = FALSE;
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].codecs, codecs, sizeof(p->offered_media[SDP_VIDEO].codecs));
@@ -8573,13 +8557,15 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
}
+ /* Search for text media definition */
} else if ((sscanf(m, "text %30d/%30d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
- (sscanf(m, "text %30d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
+ (sscanf(m, "text %30d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
text = TRUE;
- p->offered_media[SDP_TEXT].offered = TRUE;
p->notext = FALSE;
+ p->offered_media[SDP_TEXT].offered = TRUE;
numberofmediastreams++;
tportno = x;
+
/* Scan through the RTP payload types specified in a "m=" line: */
codecs = m + len;
ast_copy_string(p->offered_media[SDP_TEXT].codecs, codecs, sizeof(p->offered_media[SDP_TEXT].codecs));
@@ -8592,46 +8578,105 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_verbose("Found RTP text format %d\n", codec);
ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
}
- } 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) )) {
+ /* 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) )) {
+ 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;
udptlportno = x;
numberofmediastreams++;
- } else
+
+ if (p->t38.state != T38_ENABLED) {
+ memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));
+
+ /* Remote party offers T38, we need to update state */
+ if ((t38action == SDP_T38_ACCEPT) &&
+ (p->t38.state == T38_LOCAL_REINVITE)) {
+ change_t38_state(p, T38_ENABLED);
+ } else if ((t38action == SDP_T38_INITIATE) &&
+ p->owner && p->lastinvite) {
+ change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
+ }
+ }
+ } 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 (video) {
- 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;
+ if (process_sdp_c(value, &videohp)) {
+ processed = TRUE;
+ vhp = &videohp.hp;
}
} 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);
- return -2;
+ if (process_sdp_c(value, &texthp)) {
+ processed = TRUE;
+ thp = &texthp.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;
+ }
+ /* Text (T.140) specific scanning */
+ else if (text) {
+ if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &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 && tportno == -1)
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
@@ -8641,374 +8686,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* 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) {
- if (udptlportno > 0) {
- sin.sin_port = htons(udptlportno);
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
- struct sockaddr_in remote_address = { 0, };
- ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
- if (remote_address.sin_addr.s_addr) {
- memcpy(&sin, &remote_address, sizeof(sin));
- 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_debug(1, "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_debug(1, "Peer doesn't provide T.38 UDPTL\n");
- }
- }
-
-
- if (p->rtp) {
- if (portno > 0) {
- sin.sin_port = htons(portno);
- ast_rtp_instance_set_remote_address(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_instance_stop(p->rtp);
- if (debug)
- ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
- }
- }
- }
- /* Setup video port number, assumes we have audio */
- if (vportno != -1)
- vsin.sin_port = htons(vportno);
-
- /* 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 */
- iterator = req->sdp_start;
- while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
- char mimeSubtype[128];
- char fmtp_string[64];
- unsigned int sample_rate;
-
- 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, "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;
- }
-
- if (!strcasecmp(a, "inactive")) {
- if (sendonly == -1)
- sendonly = 2;
- continue;
- }
-
- if (!strcasecmp(a, "sendrecv")) {
- if (sendonly == -1)
- sendonly = 0;
- continue;
- }
-
- if (!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;
- ast_debug(1, "Can't read framing from SDP: %s\n", a);
- }
- }
- if (framing && p->autoframing) {
- struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
- int codec_n;
- for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
- struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
- if (!format.asterisk_format || !format.code) /* non-codec or not found */
- continue;
- if (option_debug)
- ast_log(LOG_DEBUG, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
- ast_codec_pref_setsize(pref, format.code, framing);
- }
- ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
- }
- continue;
- }
-
- if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
- /* count numbers of generations in fmtp */
- red_cp = &red_fmtp[strlen(red_fmtp)];
- strncpy(red_fmtp, a, 100);
-
- sscanf(red_cp, "%30u", &red_data_pt[red_num_gen]);
- red_cp = strtok(red_cp, "/");
- while (red_cp && red_num_gen++ < AST_RED_MAX_GENERATION) {
- sscanf(red_cp, "%30u", &red_data_pt[red_num_gen]);
- red_cp = strtok(NULL, "/");
- }
- red_cp = red_fmtp;
- continue;
- }
-
- if (sscanf(a, "fmtp: %30u %63s", &codec, fmtp_string) == 2) {
- struct ast_rtp_payload_type payload;
- unsigned int handled = 0;
-
- payload = ast_rtp_codecs_payload_lookup(&newaudiortp, codec);
- if (!payload.code) {
- /* it wasn't found, try the video rtp */
- payload = ast_rtp_codecs_payload_lookup(&newvideortp, codec);
- }
- if (payload.code && payload.asterisk_format) {
- unsigned int bit_rate;
-
- switch (payload.code) {
- case AST_FORMAT_SIREN7:
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 32000) {
- ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
- } else {
- handled = 1;
- }
- }
- break;
- case AST_FORMAT_SIREN14:
- if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
- if (bit_rate != 48000) {
- ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
- ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
- } else {
- handled = 1;
- }
- }
- break;
- }
- }
-
- if (!handled) {
- ast_debug(1, "Got unsupported a:%s in SDP offer\n", a);
- }
- continue;
- }
-
- if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
- /* We have a rtpmap to handle */
-
- if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- /* Note: should really look at the '#chans' params too */
- /* Note: This should all be done in the context of the m= above */
- if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
- if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -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++;
- } else {
- ast_rtp_codecs_payloads_unset(&newvideortp, NULL, codec);
- if (debug)
- ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
- }
- } else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
- if (p->trtp) {
- /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
- ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
- }
- } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
- if (p->trtp) {
- ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
- red_pt = codec;
- sprintf(red_fmtp, "fmtp:%d ", red_pt);
-
- if (debug)
- ast_verbose("RED submimetype has payload type: %d\n", red_pt);
- }
- } else { /* Must be audio?? */
- if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newaudiortp, NULL, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -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++;
- } else {
- ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
- if (debug)
- ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
- }
- }
- } else {
- if (debug)
- ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
- }
-
- }
- }
-
- if (udptlportno != -1) {
- if (p->t38.state != T38_ENABLED) {
- int found = 0, x;
-
- old = 0;
- memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));
-
- /* 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;
- ast_debug(3, "MaxBufferSize:%d\n", x);
- } else if ((sscanf(a, "T38MaxBitRate:%30d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%30d", &x) == 1)) {
- found = 1;
- ast_debug(3, "T38MaxBitRate: %d\n", x);
- switch (x) {
- case 14400:
- p->t38.their_parms.rate = AST_T38_RATE_14400;
- break;
- case 12000:
- p->t38.their_parms.rate = AST_T38_RATE_12000;
- break;
- case 9600:
- p->t38.their_parms.rate = AST_T38_RATE_9600;
- break;
- case 7200:
- p->t38.their_parms.rate = AST_T38_RATE_7200;
- break;
- case 4800:
- p->t38.their_parms.rate = AST_T38_RATE_4800;
- break;
- case 2400:
- p->t38.their_parms.rate = AST_T38_RATE_2400;
- break;
- }
- } else if ((sscanf(a, "T38FaxVersion:%30d", &x) == 1)) {
- found = 1;
- ast_debug(3, "FaxVersion: %d\n", x);
- p->t38.their_parms.version = x;
- } else if ((sscanf(a, "T38FaxMaxDatagram:%30d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%30d", &x) == 1)) {
- /* override the supplied value if the configuration requests it */
- if (p->t38_maxdatagram > x) {
- ast_debug(1, "Overriding T38FaxMaxDatagram '%d' with '%d'\n", x, p->t38_maxdatagram);
- x = p->t38_maxdatagram;
- }
- found = 1;
- ast_debug(3, "FaxMaxDatagram: %d\n", x);
- ast_udptl_set_far_max_datagram(p->udptl, x);
- } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
- found = 1;
- if (sscanf(a, "T38FaxFillBitRemoval:%30d", &x) == 1) {
- ast_debug(3, "FillBitRemoval: %d\n", x);
- if (x == 1) {
- p->t38.their_parms.fill_bit_removal = TRUE;
- }
- } else {
- ast_debug(3, "FillBitRemoval\n");
- p->t38.their_parms.fill_bit_removal = TRUE;
- }
- } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
- found = 1;
- if (sscanf(a, "T38FaxTranscodingMMR:%30d", &x) == 1) {
- ast_debug(3, "Transcoding MMR: %d\n", x);
- if (x == 1) {
- p->t38.their_parms.transcoding_mmr = TRUE;
- }
- } else {
- ast_debug(3, "Transcoding MMR\n");
- p->t38.their_parms.transcoding_mmr = TRUE;
- }
- } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
- found = 1;
- if (sscanf(a, "T38FaxTranscodingJBIG:%30d", &x) == 1) {
- ast_debug(3, "Transcoding JBIG: %d\n", x);
- if (x == 1) {
- p->t38.their_parms.transcoding_jbig = TRUE;
- }
- } else {
- ast_debug(3, "Transcoding JBIG\n");
- p->t38.their_parms.transcoding_jbig = TRUE;
- }
- } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
- found = 1;
- ast_debug(3, "RateManagement: %s\n", s);
- if (!strcasecmp(s, "localTCF"))
- p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_LOCAL_TCF;
- else if (!strcasecmp(s, "transferredTCF"))
- p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF;
- } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
- found = 1;
- ast_debug(3, "UDP EC: %s\n", s);
- if (!strcasecmp(s, "t38UDPRedundancy")) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
- } else if (!strcasecmp(s, "t38UDPFEC")) {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
- } else {
- ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
- }
- }
- }
-
- /* Remote party offers T38, we need to update state */
- if ((t38action == SDP_T38_ACCEPT) &&
- (p->t38.state == T38_LOCAL_REINVITE)) {
- change_t38_state(p, T38_ENABLED);
- } else if ((t38action == SDP_T38_INITIATE) &&
- p->owner && p->lastinvite) {
- change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
- }
- }
- } else {
+ if (udptlportno == -1) {
change_t38_state(p, T38_DISABLED);
}
+
/* Now gather all of the codecs that we are asked for: */
ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
@@ -9029,7 +8711,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));
}
-
if (debug) {
struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE);
struct ast_str *s2 = ast_str_alloca(SIPBUFSIZE);
@@ -9090,25 +8771,79 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
}
- /* Setup audio port number */
- if (p->rtp && sin.sin_port) {
- ast_rtp_instance_set_remote_address(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));
+ /* Setup audio address and 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_instance_set_remote_address(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_instance_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_instance_set_remote_address(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));
+ /* Setup video address and 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_instance_set_remote_address(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_instance_stop(p->vrtp);
+ if (debug)
+ ast_verbose("Peer doesn't provide video\n");
+ }
}
- /* Setup text port number */
- if (p->trtp && tsin.sin_port) {
- ast_rtp_instance_set_remote_address(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));
+ /* Setup text address and port */
+ if (p->trtp) {
+ if (tportno > 0) {
+ tsin.sin_family = AF_INET;
+ tsin.sin_port = htons(tportno);
+ memcpy(&tsin.sin_addr, thp->h_addr, sizeof(tsin.sin_addr));
+ ast_rtp_instance_set_remote_address(p->trtp, &tsin);
+ if (debug)
+ ast_verbose("Peer T.140 RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
+ } else {
+ ast_rtp_instance_stop(p->trtp);
+ if (debug)
+ ast_verbose("Peer doesn't provide T.140\n");
+ }
+ }
+ /* Setup image address and port */
+ if (p->udptl) {
+ if (udptlportno > 0) {
+ isin.sin_family = AF_INET;
+ isin.sin_port = htons(udptlportno);
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
+ struct sockaddr_in remote_address = { 0, };
+ ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
+ if (remote_address.sin_addr.s_addr) {
+ memcpy(&isin, &remote_address, sizeof(isin));
+ 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));
+ }
+ }
+ }
+ ast_udptl_set_peer(p->udptl, &isin);
+ if (debug)
+ ast_debug(1,"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_debug(1, "Peer doesn't provide T.38 UDPTL\n");
+ }
}
/* Ok, we're going with this offer */
@@ -9180,6 +8915,396 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
return 0;
}
+static int process_sdp_o(const char *o, struct sip_pvt *p)
+{
+ char *o_copy;
+ char *token;
+ int64_t rua_version;
+
+ /* Store the SDP version number of remote UA. This will allow us to
+ distinguish between session modifications and session refreshes. If
+ the remote UA does not send an incremented SDP version number in a
+ subsequent RE-INVITE then that means its not changing media session.
+ The RE-INVITE may have been sent to update connected party, remote
+ target or to refresh the session (Session-Timers). Asterisk must not
+ change media session and increment its own version number in answer
+ SDP in this case. */
+
+ if (ast_strlen_zero(o)) {
+ ast_log(LOG_WARNING, "SDP syntax error. SDP without an o= line\n");
+ return FALSE;
+ }
+
+ o_copy = ast_strdupa(o);
+ token = strsep(&o_copy, " "); /* Skip username */
+ if (!o_copy) {
+ ast_log(LOG_WARNING, "SDP syntax error in o= line username\n");
+ return FALSE;
+ }
+ token = strsep(&o_copy, " "); /* Skip session-id */
+ if (!o_copy) {
+ ast_log(LOG_WARNING, "SDP syntax error in o= line session-id\n");
+ return FALSE;
+ }
+ token = strsep(&o_copy, " "); /* Version */
+ if (!o_copy) {
+ ast_log(LOG_WARNING, "SDP syntax error in o= line\n");
+ return FALSE;
+ }
+ if (!sscanf(token, "%30" SCNd64, &rua_version)) {
+ ast_log(LOG_WARNING, "SDP syntax error in o= line version\n");
+ return FALSE;
+ }
+
+ /* we need to check the SDP version number the other end sent us;
+ * our rules for deciding what to accept are a bit complex.
+ *
+ * 1) if 'ignoresdpversion' has been set for this dialog, then
+ * we will just accept whatever they sent and assume it is
+ * a modification of the session, even if it is not
+ * 2) otherwise, if this is the first SDP we've seen from them
+ * we accept it
+ * 3) otherwise, if the new SDP version number is higher than the
+ * old one, we accept it
+ * 4) otherwise, if this SDP is in response to us requesting a switch
+ * to T.38, we accept the SDP, but also generate a warning message
+ * that this peer should have the 'ignoresdpversion' option set,
+ * because it is not following the SDP offer/answer RFC; if we did
+ * not request a switch to T.38, then we stop parsing the SDP, as it
+ * has not changed from the previous version
+ */
+
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION) ||
+ (p->sessionversion_remote < 0) ||
+ (p->sessionversion_remote < rua_version)) {
+ p->sessionversion_remote = rua_version;
+ p->session_modify = TRUE;
+ } else {
+ if (p->t38.state == T38_LOCAL_REINVITE) {
+ p->sessionversion_remote = rua_version;
+ p->session_modify = TRUE;
+ ast_log(LOG_WARNING, "Call %s responded to our T.38 reinvite without changing SDP version; 'ignoresdpversion' should be set for this peer.\n", p->callid);
+ } else {
+ p->session_modify = FALSE;
+ ast_debug(2, "Call %s responded to our reinvite without changing SDP version; ignoring SDP.\n", p->callid);
+ return FALSE;
+ }
+ }
+
+ return TRUE;
+}
+
+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_codecs *newaudiortp, int *last_rtpmap_codec)
+{
+ int found = FALSE;
+ int codec;
+ char mimeSubtype[128];
+ char fmtp_string[64];
+ unsigned int sample_rate;
+ int debug = sip_debug_test_pvt(p);
+
+ if (!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;
+ ast_debug(1, "Can't read framing from SDP: %s\n", a);
+ }
+ }
+ if (framing && p->autoframing) {
+ struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
+ int codec_n;
+ for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
+ struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
+ if (!format.asterisk_format || !format.code) /* non-codec or not found */
+ continue;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
+ ast_codec_pref_setsize(pref, format.code, framing);
+ }
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
+ }
+ found = TRUE;
+ } else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
+ /* We have a rtpmap to handle */
+ if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
+ if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -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 {
+ ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
+ if (debug)
+ ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ } else {
+ if (debug)
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ } else if (sscanf(a, "fmtp: %30u %63s", &codec, fmtp_string) == 2) {
+ struct ast_rtp_payload_type payload;
+
+ payload = ast_rtp_codecs_payload_lookup(newaudiortp, codec);
+ if (payload.code && payload.asterisk_format) {
+ unsigned int bit_rate;
+
+ switch (payload.code) {
+ case AST_FORMAT_SIREN7:
+ if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
+ if (bit_rate != 32000) {
+ ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
+ ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
+ } else {
+ found = TRUE;
+ }
+ }
+ break;
+ case AST_FORMAT_SIREN14:
+ if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
+ if (bit_rate != 48000) {
+ ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
+ ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
+ } else {
+ found = TRUE;
+ }
+ }
+ break;
+ }
+ }
+ }
+
+ return found;
+}
+
+static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec)
+{
+ int found = FALSE;
+ int codec;
+ char mimeSubtype[128];
+ unsigned int sample_rate;
+ int debug = sip_debug_test_pvt(p);
+
+ if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
+ /* We have a rtpmap to handle */
+ if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
+ /* Note: should really look at the '#chans' params too */
+ if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
+ if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -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 {
+ ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
+ if (debug)
+ ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ }
+ } else {
+ if (debug)
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ }
+
+ return found;
+}
+
+static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec)
+{
+ int found = FALSE;
+ int codec;
+ char mimeSubtype[128];
+ unsigned int sample_rate;
+ char *red_cp;
+ int debug = sip_debug_test_pvt(p);
+
+ if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
+ /* We have a rtpmap to handle */
+ if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
+ if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
+ if (p->trtp) {
+ /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
+ ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
+ found = TRUE;
+ }
+ } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
+ if (p->trtp) {
+ ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
+ sprintf(red_fmtp, "fmtp:%d ", codec);
+ if (debug)
+ ast_verbose("RED submimetype has payload type: %d\n", codec);
+ found = TRUE;
+ }
+ }
+ } else {
+ if (debug)
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
+ }
+ } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
+ /* count numbers of generations in fmtp */
+ red_cp = &red_fmtp[strlen(red_fmtp)];
+ strncpy(red_fmtp, a, 100);
+
+ sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
+ red_cp = strtok(red_cp, "/");
+ while (red_cp && (*red_num_gen)++ < AST_RED_MAX_GENERATION) {
+ sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
+ red_cp = strtok(NULL, "/");
+ }
+ red_cp = red_fmtp;
+ found = TRUE;
+ }
+
+ return found;
+}
+
+static int process_sdp_a_image(const char *a, struct sip_pvt *p)
+{
+ int found = FALSE;
+ char s[256];
+ int x;
+
+ if ((sscanf(a, "T38FaxMaxBuffer:%30d", &x) == 1)) {
+ ast_debug(3, "MaxBufferSize:%d\n", x);
+ found = TRUE;
+ } else if ((sscanf(a, "T38MaxBitRate:%30d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%30d", &x) == 1)) {
+ ast_debug(3, "T38MaxBitRate: %d\n", x);
+ switch (x) {
+ case 14400:
+ p->t38.their_parms.rate = AST_T38_RATE_14400;
+ break;
+ case 12000:
+ p->t38.their_parms.rate = AST_T38_RATE_12000;
+ break;
+ case 9600:
+ p->t38.their_parms.rate = AST_T38_RATE_9600;
+ break;
+ case 7200:
+ p->t38.their_parms.rate = AST_T38_RATE_7200;
+ break;
+ case 4800:
+ p->t38.their_parms.rate = AST_T38_RATE_4800;
+ break;
+ case 2400:
+ p->t38.their_parms.rate = AST_T38_RATE_2400;
+ break;
+ }
+ found = TRUE;
+ } else if ((sscanf(a, "T38FaxVersion:%30d", &x) == 1)) {
+ ast_debug(3, "FaxVersion: %d\n", x);
+ p->t38.their_parms.version = x;
+ found = TRUE;
+ } else if ((sscanf(a, "T38FaxMaxDatagram:%30d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%30d", &x) == 1)) {
+ /* override the supplied value if the configuration requests it */
+ if (p->t38_maxdatagram > x) {
+ ast_debug(1, "Overriding T38FaxMaxDatagram '%d' with '%d'\n", x, p->t38_maxdatagram);
+ x = p->t38_maxdatagram;
+ }
+ ast_debug(3, "FaxMaxDatagram: %d\n", x);
+ ast_udptl_set_far_max_datagram(p->udptl, x);
+ found = TRUE;
+ } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
+ if (sscanf(a, "T38FaxFillBitRemoval:%30d", &x) == 1) {
+ ast_debug(3, "FillBitRemoval: %d\n", x);
+ if (x == 1) {
+ p->t38.their_parms.fill_bit_removal = TRUE;
+ }
+ } else {
+ ast_debug(3, "FillBitRemoval\n");
+ p->t38.their_parms.fill_bit_removal = TRUE;
+ }
+ found = TRUE;
+ } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
+ if (sscanf(a, "T38FaxTranscodingMMR:%30d", &x) == 1) {
+ ast_debug(3, "Transcoding MMR: %d\n", x);
+ if (x == 1) {
+ p->t38.their_parms.transcoding_mmr = TRUE;
+ }
+ } else {
+ ast_debug(3, "Transcoding MMR\n");
+ p->t38.their_parms.transcoding_mmr = TRUE;
+ }
+ found = TRUE;
+ } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
+ if (sscanf(a, "T38FaxTranscodingJBIG:%30d", &x) == 1) {
+ ast_debug(3, "Transcoding JBIG: %d\n", x);
+ if (x == 1) {
+ p->t38.their_parms.transcoding_jbig = TRUE;
+ }
+ } else {
+ ast_debug(3, "Transcoding JBIG\n");
+ p->t38.their_parms.transcoding_jbig = TRUE;
+ }
+ found = TRUE;
+ } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
+ ast_debug(3, "RateManagement: %s\n", s);
+ if (!strcasecmp(s, "localTCF"))
+ p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_LOCAL_TCF;
+ else if (!strcasecmp(s, "transferredTCF"))
+ p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF;
+ found = TRUE;
+ } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
+ ast_debug(3, "UDP EC: %s\n", s);
+ if (!strcasecmp(s, "t38UDPRedundancy")) {
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
+ } else if (!strcasecmp(s, "t38UDPFEC")) {
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
+ } else {
+ ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
+ }
+ found = TRUE;
+ }
+
+ return found;
+}
+
+
#ifdef LOW_MEMORY
static void ts_ast_rtp_destroy(void *data)
{