[svn-commits] kpfleming: branch kpfleming/sdp-parsing-unbreakage r229637 - /team/kpfleming/...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Nov 12 07:25:49 CST 2009
Author: kpfleming
Date: Thu Nov 12 07:25:45 2009
New Revision: 229637
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=229637
Log:
restructure session state change logic so it actually works
convert sendonly magic numbers to enum
more coding guidelines fixes
Modified:
team/kpfleming/sdp-parsing-unbreakage/channels/chan_sip.c
Modified: team/kpfleming/sdp-parsing-unbreakage/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/kpfleming/sdp-parsing-unbreakage/channels/chan_sip.c?view=diff&rev=229637&r1=229636&r2=229637
==============================================================================
--- team/kpfleming/sdp-parsing-unbreakage/channels/chan_sip.c (original)
+++ team/kpfleming/sdp-parsing-unbreakage/channels/chan_sip.c Thu Nov 12 07:25:45 2009
@@ -1803,6 +1803,13 @@
static int sip_sipredirect(struct sip_pvt *p, const char *dest);
/*--- Codec handling / SDP */
+enum media_session_mode {
+ MEDIA_SENDRECV = 0,
+ MEDIA_SENDONLY,
+ MEDIA_RECVONLY,
+ MEDIA_INACTIVE,
+};
+
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 char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value);
@@ -1810,7 +1817,7 @@
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_mode(const char *a, enum media_session_mode *mode);
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_text(const char *a, struct sip_pvt *p, struct ast_rtp *newtextrtp, int *last_rtpmap_codec);
@@ -6836,9 +6843,9 @@
/* We only want the m and c lines for audio */
for (m = get_sdp_iterate(&miterator, req, "m"); !ast_strlen_zero(m); m = get_sdp_iterate(&miterator, req, "m")) {
if ((media == SDP_AUDIO && ((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))) ||
- (media == SDP_VIDEO && ((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, "audio %30d RTP/AVP %n", &x, &len) == 1 && len > 0))) ||
+ (media == SDP_VIDEO && ((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)))) {
/* See if there's a c= line for this media stream.
* XXX There is no guarantee that we'll be grabbing the c= line for this
* particular media stream here. However, this is the same logic used in process_sdp.
@@ -6915,8 +6922,8 @@
int codec;
/* Others */
- int sendonly = -1;
- int vsendonly = -1;
+ enum media_session_mode amode = MEDIA_SENDRECV;
+ enum media_session_mode vmode = MEDIA_SENDRECV;
int numberofports;
int numberofmediastreams = 0;
int last_rtpmap_codec = 0;
@@ -6992,9 +6999,9 @@
}
break;
case 'a':
- if (process_sdp_a_sendonly(value, &sendonly)) {
+ if (process_sdp_a_mode(value, &amode)) {
processed = TRUE;
- vsendonly = sendonly;
+ vmode = amode;
} 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)) {
@@ -7101,18 +7108,6 @@
if (p->t38.state != T38_ENABLED) {
memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));
-
- /* KPF: these state changes should not occur until we decide whether
- we are actually going to accept this offer or not
- */
- /* 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);
@@ -7154,13 +7149,13 @@
break;
case 'a':
if (audio) {
- if (process_sdp_a_sendonly(value, &sendonly)) {
+ if (process_sdp_a_mode(value, &amode)) {
processed = TRUE;
} else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec)) {
processed = TRUE;
}
} else if (video) {
- if (process_sdp_a_sendonly(value, &vsendonly)) {
+ if (process_sdp_a_mode(value, &vmode)) {
processed = TRUE;
} else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec)) {
processed = TRUE;
@@ -7187,7 +7182,7 @@
}
/* Sanity checks */
- if (aportno == -1 && vportno == -1 && iportno == -1 && tportno == -1) {
+ if (aportno == -1 && vportno == -1 && iportno == -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;
@@ -7196,10 +7191,6 @@
if (numberofmediastreams > 3) {
/* We have too many fax, audio and/or video and/or text media streams, fail this offer */
return -3;
- }
-
- if (iportno == -1) {
- change_t38_state(p, T38_DISABLED);
}
/* Now gather all of the codecs that we are asked for: */
@@ -7227,47 +7218,17 @@
ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, apeernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0));
}
- if (!newjointcapability) {
- /* If T.38 was not negotiated either, totally bail out... */
- if ((p->t38.state == T38_DISABLED) || !iportno) {
- ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
- /* Do NOT Change current setting */
- return -1;
- } else {
- ast_debug(3, "Have T.38 but no audio codecs, accepting offer anyway\n");
- /* KPF: except we don't actually finish processing the offer, so the call is broken */
- return 0;
- }
- }
-
- /* 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->peercapability = newpeercapability; /* The other sides capability in latest offer */
- p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
-
- 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);
- if (newnoncodeccapability & AST_RTP_DTMF) {
- /* XXX Would it be reasonable to drop the DSP at this point? XXX */
- ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
- /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
- ast_rtp_setdtmf(p->rtp, 1);
- ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- } else {
- ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
- }
- }
-
+
+ if (!newjointcapability && (aportno != -1)) {
+ ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
+ return -1;
+ }
+
+ /* We are now ready to change the sip session with the offered codecs, since
+ they are acceptable */
/* Setup audio address and port */
if (p->rtp) {
- if (aportno > 0) {
+ if (aportno != -1) {
struct sockaddr_in sin;
sin.sin_family = AF_INET;
@@ -7277,6 +7238,23 @@
if (debug) {
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
}
+ p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
+ p->peercapability = newpeercapability; /* The other sides capability in latest offer */
+ p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
+
+ ast_rtp_pt_copy(p->rtp, newaudiortp);
+ if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
+ ast_clear_flag(&p->flags[0], SIP_DTMF);
+ if (newnoncodeccapability & AST_RTP_DTMF) {
+ /* XXX Would it be reasonable to drop the DSP at this point? XXX */
+ ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
+ /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
+ ast_rtp_setdtmf(p->rtp, 1);
+ ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ } else {
+ ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
+ }
+ }
} else {
ast_rtp_stop(p->rtp);
if (debug) {
@@ -7284,10 +7262,9 @@
}
}
}
-
/* Setup video address and port */
if (p->vrtp) {
- if (vportno > 0) {
+ if (vportno != -1) {
struct sockaddr_in sin;
sin.sin_family = AF_INET;
@@ -7297,6 +7274,7 @@
if (debug) {
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
}
+ ast_rtp_pt_copy(p->vrtp, newvideortp);
} else {
ast_rtp_stop(p->vrtp);
if (debug) {
@@ -7304,10 +7282,9 @@
}
}
}
-
/* Setup text address and port */
if (p->trtp) {
- if (tportno > 0) {
+ if (tportno != -1) {
struct sockaddr_in sin;
sin.sin_family = AF_INET;
@@ -7317,6 +7294,7 @@
if (debug) {
ast_verbose("Peer T.140 RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
}
+ ast_rtp_pt_copy(p->trtp, newtextrtp);
} else {
ast_rtp_stop(p->trtp);
if (debug) {
@@ -7326,13 +7304,14 @@
}
/* Setup image address and port */
if (p->udptl) {
- if (iportno > 0) {
+ if (iportno != -1) {
struct sockaddr_in sin;
sin.sin_family = AF_INET;
sin.sin_port = htons(iportno);
if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
struct sockaddr_in remote_address = { 0, };
+
ast_rtp_get_peer(p->rtp, &remote_address);
if (remote_address.sin_addr.s_addr) {
memcpy(&sin, &remote_address, sizeof(sin));
@@ -7345,12 +7324,26 @@
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));
}
+ /* 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 {
+ change_t38_state(p, T38_DISABLED);
ast_udptl_stop(p->udptl);
if (debug) {
ast_debug(1, "Peer doesn't want T.38 UDPTL\n");
}
}
+ }
+
+ if ((aportno == -1) && (p->t38.state != T38_DISABLED)) {
+ ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n");
+ return 0;
}
/* Ok, we're going with this offer */
@@ -7375,7 +7368,7 @@
ast_set_write_format(p->owner, p->owner->writeformat);
}
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && ahp && ahp->h_addr && (!sendonly || sendonly == -1)) {
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && ahp && ahp->h_addr && (amode == MEDIA_SENDRECV)) {
ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
/* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame);
@@ -7393,13 +7386,13 @@
sip_peer_hold(p, FALSE);
}
ast_clear_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
- } else if ((ahp && !ahp->h_addr) || (sendonly && sendonly != -1)) {
+ } else if ((ahp && !ahp->h_addr) || (amode != MEDIA_SENDRECV)) {
int already_on_hold = ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD);
ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
S_OR(p->mohsuggest, NULL),
!ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
- if (sendonly) {
+ if (amode != MEDIA_SENDRECV) {
ast_rtp_stop(p->rtp);
}
/* RTCP needs to go ahead, even if we're on hold!!! */
@@ -7415,10 +7408,10 @@
p->owner->name,
p->owner->uniqueid);
}
- if (sendonly == 1) {
+ if (amode == MEDIA_SENDONLY) {
/* One directional hold (sendonly/recvonly) */
ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR);
- } else if (sendonly == 2) {
+ } else if (amode == MEDIA_INACTIVE) {
/* Inactive stream */
ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE);
} else {
@@ -7530,24 +7523,19 @@
return FALSE;
}
-static int process_sdp_a_sendonly(const char *a, int *sendonly)
-{
- int found = FALSE;
-
+static int process_sdp_a_mode(const char *a, enum media_session_mode *mode)
+{
if (!strcasecmp(a, "sendonly")) {
- if (*sendonly == -1)
- *sendonly = 1;
- found = TRUE;
+ *mode = MEDIA_SENDONLY;
+ return TRUE;
} else if (!strcasecmp(a, "inactive")) {
- if (*sendonly == -1)
- *sendonly = 2;
- found = TRUE;
+ *mode = MEDIA_INACTIVE;
+ return TRUE;
} else if (!strcasecmp(a, "sendrecv")) {
- if (*sendonly == -1)
- *sendonly = 0;
- found = TRUE;
- }
- return found;
+ *mode = MEDIA_SENDRECV;
+ return TRUE;
+ }
+ return FALSE;
}
static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp *newaudiortp, int *last_rtpmap_codec)
@@ -7574,10 +7562,13 @@
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 */
+ if (!format) {
+ /* non-codec or not found */
continue;
- if (option_debug)
+ }
+ 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);
@@ -7588,19 +7579,20 @@
if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
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)
+ 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_unset_m_type(newaudiortp, codec);
- if (debug)
+ 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 (debug) {
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
}
@@ -7620,20 +7612,21 @@
/* Note: should really look at the '#chans' params too */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
if (ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
- if (debug)
+ 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_unset_m_type(newvideortp, codec);
- if (debug)
+ 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 (debug) {
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
}
@@ -7657,9 +7650,8 @@
found = TRUE;
}
}
- } else {
- if (debug)
- ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
+ } else if (debug) {
+ ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
}
@@ -7673,7 +7665,7 @@
int x;
if ((sscanf(a, "T38FaxMaxBuffer:%30d", &x) == 1)) {
- ast_debug(3, "MaxBufferSize:%d\n", x);
+ 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);
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