[svn-commits] branch oej/sdpcleanup r25926 - in
/team/oej/sdpcleanup: ./ channels/ include/...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue May 9 01:28:27 MST 2006
Author: oej
Date: Tue May 9 03:28:27 2006
New Revision: 25926
URL: http://svn.digium.com/view/asterisk?rev=25926&view=rev
Log:
- Cleanup process_sdp
- Preparing for fmtp: support
- Move some definitions from rtp.c to rtp.h since they are needed outside of rtp.c
Modified:
team/oej/sdpcleanup/channels/chan_sip.c
team/oej/sdpcleanup/include/asterisk/rtp.h
team/oej/sdpcleanup/rtp.c
Modified: team/oej/sdpcleanup/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/sdpcleanup/channels/chan_sip.c?rev=25926&r1=25925&r2=25926&view=diff
==============================================================================
--- team/oej/sdpcleanup/channels/chan_sip.c (original)
+++ team/oej/sdpcleanup/channels/chan_sip.c Tue May 9 03:28:27 2006
@@ -3027,6 +3027,7 @@
struct ast_variable *v = NULL;
int fmt;
int what;
+ int needvideo = 0;
ast_mutex_unlock(&i->lock);
/* Don't hold a sip pvt lock while we allocate a channel */
@@ -3045,6 +3046,10 @@
what = i->capability;
else
what = global_capability;
+ needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK;
+ if (option_debug && needvideo)
+ ast_log(LOG_DEBUG, "This call needs video! HOLLYWOOD next!\n");
+
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
@@ -3065,7 +3070,7 @@
tmp->fds[0] = ast_rtp_fd(i->rtp);
tmp->fds[1] = ast_rtcp_fd(i->rtp);
}
- if (i->vrtp) {
+ if (needvideo && i->vrtp) {
tmp->fds[2] = ast_rtp_fd(i->vrtp);
tmp->fds[3] = ast_rtcp_fd(i->vrtp);
}
@@ -3726,26 +3731,40 @@
char host[258];
char iabuf[INET_ADDRSTRLEN];
int len = -1;
- int portno = -1;
- int vportno = -1;
+ int portno = -1; /*!< RTP Audio port number */
+ int vportno = -1; /*!< RTP Video port number */
int peercapability, peernoncodeccapability;
int vpeercapability=0, vpeernoncodeccapability=0;
struct sockaddr_in sin;
+ struct sockaddr_in vsin; /*!< Video socket address */
const char *codecs;
- struct hostent *hp;
- struct ast_hostent ahp;
+ 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 = 0;
- int x,y;
+ int x;
+ int numberofports;
int debug=sip_debug_test_pvt(p);
struct ast_channel *bridgepeer = NULL;
-
+ struct ast_rtp newaudiortp, newvideortp;
+ int newjointcapability;
+ int newpeercapability;
+ int newnoncodeccapability;
+ int numberofmediastreams = 0;
+
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1;
}
+
+ ast_rtp_pt_default(&newaudiortp);
+ if (p->vrtp)
+ ast_rtp_pt_default(&newvideortp);
+
/* Update our last rtprx when we receive an SDP, too */
time(&p->lastrtprx);
@@ -3756,6 +3775,7 @@
ast_log(LOG_NOTICE, "Content is '%s', not 'application/sdp'\n", get_header(req, "Content-Type"));
return -1;
}
+ /* Try to find first media stream */
m = get_sdp(req, "m");
destiterator = 0;
c = get_sdp_iterate(&destiterator, req, "c");
@@ -3768,21 +3788,30 @@
return -1;
}
/* XXX This could block for a long time, and block the main thread! XXX */
- hp = ast_gethostbyname(host, &ahp);
+ hp = ast_gethostbyname(host, &audiohp);
if (!hp) {
ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
return -1;
}
iterator = 0;
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') {
- int found = 0;
- if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 2) ||
+ int audio = FALSE;
+ numberofmediastreams++;
+
+ if (p->vrtp)
+ ast_rtp_pt_clear(&newvideortp); /* Must be cleared in case no m=video line exists */
+ numberofports = 1;
+ if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
- found = 1;
+ audio = TRUE;
+ /* Found audio stream in this media definition */
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
- ast_rtp_pt_clear(p->rtp);
+ ast_rtp_pt_clear(&newaudiortp);
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -3790,14 +3819,11 @@
}
if (debug)
ast_verbose("Found RTP audio format %d\n", codec);
- ast_rtp_set_m_type(p->rtp, codec);
+ ast_rtp_set_m_type(&newaudiortp, codec);
}
- }
- if (p->vrtp)
- ast_rtp_pt_clear(p->vrtp); /* Must be cleared in case no m=video line exists */
-
- if (p->vrtp && (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
- found = 1;
+ } else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
+ (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
+ /* If it is not audio - is it video ? */
ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
@@ -3807,110 +3833,139 @@
return -1;
}
if (debug)
- ast_verbose("Found RTP video format %d\n", codec);
- ast_rtp_set_m_type(p->vrtp, codec);
- }
- }
- if (!found )
- ast_log(LOG_WARNING, "Unknown SDP media type in offer: %s\n", m);
- }
- if (portno == -1 && vportno == -1) {
- /* No acceptable offer found in SDP */
- return -2;
- }
- /* Check for Media-description-level-address for audio */
- if (pedanticsipchecking) {
- 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 */
- hp = ast_gethostbyname(host, &ahp);
- if (!hp) {
- ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c);
+ ast_verbose("Found RTP video format %d\n", codec);
+ ast_rtp_set_m_type(&newvideortp, codec);
+ }
+ } else
+ ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
+ 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 */
+ if (pedanticsipchecking) {
+ 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 */
+ if (!audio && !(hp = ast_gethostbyname(host, &audiohp)))
+ ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
+ else if (!(vhp = ast_gethostbyname(host, &videohp)))
+ ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
}
}
- }
- }
+
+ }
+ }
+ if (portno == -1 && vportno == -1)
+ /* No acceptable offer found in SDP - we have no ports */
+ /* Do not change RTP or VRTP if this is a re-invite */
+ return -2;
+
+ if (numberofmediastreams > 2)
+ /* We have too many 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 audio port number */
sin.sin_port = htons(portno);
+ /* Setup video port number */
+ vsin.sin_port = htons(vportno);
+
if (p->rtp && sin.sin_port) {
ast_rtp_set_peer(p->rtp, &sin);
- if (debug) {
+ if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
- }
- }
- /* Check for Media-description-level-address for video */
- if (pedanticsipchecking) {
- 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 */
- hp = ast_gethostbyname(host, &ahp);
- if (!hp) {
- ast_log(LOG_WARNING, "Unable to lookup host in secondary c= line, '%s'\n", c);
- }
- }
- }
- }
+ }
+
/* Setup video port number */
- sin.sin_port = htons(vportno);
- if (p->vrtp && sin.sin_port) {
- ast_rtp_set_peer(p->vrtp, &sin);
- if (debug) {
- ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
- }
+ 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(iabuf,sizeof(iabuf), vsin.sin_addr), ntohs(vsin.sin_port));
}
/* 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 = 0;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
if (!strcasecmp(a, "sendonly")) {
- sendonly=1;
+ sendonly = 1;
continue;
- }
- if (!strcasecmp(a, "sendrecv")) {
- sendonly=0;
- }
- if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
+ } else if (!strcasecmp(a, "sendrecv")) {
+ sendonly = 0;
+ continue;
+ } else if (!strcasecmp(a, "inactive")) {
+ /* Inactive media streams: Not supported */
+ if (debug)
+ ast_verbose("Got unsupported a:inactive in SDP offer \n");
+ continue;
+ } 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");
+ continue;
+ } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
+ /* Video stuff: Not supported */
+ if (debug)
+ ast_verbose("Got unsupported a:maxprate in SDP offer \n");
+ continue;
+ } 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");
+ continue;
+ } else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
+ if (debug)
+ ast_verbose("Got unsupported a:ptime in SDP offer \n");
+ continue;
+ } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
+ continue;
+ /* We have a rtpmap to handle */
if (debug)
- ast_verbose("Found description format %s\n", mimeSubtype);
+ ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
+
/* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype);
+ ast_rtp_set_rtpmap_type(&newaudiortp, codec, "audio", mimeSubtype);
if (p->vrtp)
- ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype);
- }
-
- /* Now gather all of the codecs that were asked for: */
- ast_rtp_get_current_formats(p->rtp,
- &peercapability, &peernoncodeccapability);
+ ast_rtp_set_rtpmap_type(&newvideortp, codec, "video", mimeSubtype);
+ }
+
+ /* Now gather all of the codecs that we are asked for: */
+ ast_rtp_get_current_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
if (p->vrtp)
- ast_rtp_get_current_formats(p->vrtp,
- &vpeercapability, &vpeernoncodeccapability);
- p->jointcapability = p->capability & (peercapability | vpeercapability);
- p->peercapability = (peercapability | vpeercapability);
- p->noncodeccapability = noncodeccapability & peernoncodeccapability;
-
+ ast_rtp_get_current_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
+
+ newjointcapability = p->capability & (peercapability | vpeercapability);
+ newpeercapability = (peercapability | vpeercapability);
+ newnoncodeccapability = noncodeccapability & peernoncodeccapability;
+
+ /* THIS SHOULD MOVE TO AFTER ERROR CHECKING */
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
- if (p->noncodeccapability & AST_RTP_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);
} else {
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
}
}
-
+
if (debug) {
/* shame on whoever coded this.... */
const unsigned slen=512;
@@ -3918,34 +3973,60 @@
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
ast_getformatname_multiple(s1, slen, p->capability),
- ast_getformatname_multiple(s2, slen, peercapability),
+ ast_getformatname_multiple(s2, slen, newpeercapability),
ast_getformatname_multiple(s3, slen, vpeercapability),
- ast_getformatname_multiple(s4, slen, p->jointcapability));
+ ast_getformatname_multiple(s4, slen, newjointcapability));
ast_verbose("Non-codec capabilities: us - %s, peer - %s, combined - %s\n",
ast_rtp_lookup_mime_multiple(s1, slen, noncodeccapability, 0),
ast_rtp_lookup_mime_multiple(s2, slen, peernoncodeccapability, 0),
- ast_rtp_lookup_mime_multiple(s3, slen, p->noncodeccapability, 0));
- }
- if (!p->jointcapability) {
+ ast_rtp_lookup_mime_multiple(s3, slen, newnoncodeccapability, 0));
+ }
+ if (!newjointcapability) {
ast_log(LOG_NOTICE, "No compatible codecs!\n");
+ /* Do NOT Change current setting */
return -1;
}
if (!p->owner) /* There's no open channel owning us */
return 0;
+
+ /* Ok, we're going with this offer */
+ if (debug && option_debug)
+ ast_log(LOG_DEBUG, "Ok, we're accepting this SDP offer! \n");
+
+ p->jointcapability = newjointcapability;
+ p->peercapability = newpeercapability;
+ p->noncodeccapability = newnoncodeccapability;
+
+ /* We are now ready to change p->rtp and p->vrtp with the offered codecs, since
+ they are acceptable */
+ {
+ int i;
+ /* Copy payload types from source to destination */
+ for (i=0; i < MAX_RTP_PT; ++i) {
+ p->rtp->current_RTP_PT[i].isAstFormat = newaudiortp.current_RTP_PT[i].isAstFormat;
+ p->rtp->current_RTP_PT[i].code = newaudiortp.current_RTP_PT[i].code;
+ if (p->vrtp) {
+ p->vrtp->current_RTP_PT[i].isAstFormat = newvideortp.current_RTP_PT[i].isAstFormat;
+ p->vrtp->current_RTP_PT[i].code = newvideortp.current_RTP_PT[i].code;
+ }
+ }
+ }
if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
const unsigned slen=512;
char s1[slen], s2[slen];
- ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n",
+ if (debug)
+ ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n",
ast_getformatname_multiple(s1, slen, p->jointcapability),
ast_getformatname_multiple(s2, slen, p->owner->nativeformats));
p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
- if ((bridgepeer=ast_bridged_channel(p->owner))) {
+
+ if ((bridgepeer = ast_bridged_channel(p->owner))) {
/* We have a bridge */
/* Turn on/off music on hold if we are holding/unholding */
if (sin.sin_addr.s_addr && !sendonly) {
@@ -3959,12 +4040,15 @@
ast_moh_start(bridgepeer, NULL);
if (sendonly)
ast_rtp_stop(p->rtp);
+ /* RTCP needs to go ahead, even if we're on hold!!! */
+
/* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame);
}
}
/* Manager Hold and Unhold events must be generated, if necessary */
+ /* XXX Support for sendonly/recvonly needs to be fixed !!! */
if (sin.sin_addr.s_addr && !sendonly) {
append_history(p, "Unhold", "%s", req->data);
@@ -3993,6 +4077,7 @@
return 0;
}
+
/*! \brief Add header to SIP message */
static int add_header(struct sip_request *req, const char *var, const char *value)
@@ -4723,7 +4808,7 @@
if ((p->vrtp) &&
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
(capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
- snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate);
+ snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate);
snprintf(t, sizeof(t), "t=0 0\r\n");
ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
@@ -9850,6 +9935,7 @@
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
+ int res = 0;
if (option_debug > 3) {
int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
@@ -9885,7 +9971,7 @@
ast_setstate(p->owner, AST_STATE_RINGING);
}
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
- process_sdp(p, req);
+ res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@@ -9897,7 +9983,7 @@
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
- process_sdp(p, req);
+ res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@@ -9909,7 +9995,9 @@
sip_cancel_destroy(p);
p->authtries = 0;
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp"))
- process_sdp(p, req);
+ if (res = process_sdp(p, req) && !ast_test_flag(req, SIP_PKT_IGNORE))
+ /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
+ ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Parse contact header for continued conversation */
/* When we get 200 OK, we know which device (and IP) to contact for this call */
@@ -12370,6 +12458,9 @@
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */
return NULL;
}
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
+
if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data);
*cause = AST_CAUSE_SWITCH_CONGESTION;
Modified: team/oej/sdpcleanup/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/team/oej/sdpcleanup/include/asterisk/rtp.h?rev=25926&r1=25925&r2=25926&view=diff
==============================================================================
--- team/oej/sdpcleanup/include/asterisk/rtp.h (original)
+++ team/oej/sdpcleanup/include/asterisk/rtp.h Tue May 9 03:28:27 2006
@@ -40,13 +40,15 @@
/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
/*! DTMF (RFC2833) */
-#define AST_RTP_DTMF (1 << 0)
+#define AST_RTP_DTMF (1 << 0)
/*! 'Comfort Noise' (RFC3389) */
-#define AST_RTP_CN (1 << 1)
+#define AST_RTP_CN (1 << 1)
/*! DTMF (Cisco Proprietary) */
-#define AST_RTP_CISCO_DTMF (1 << 2)
+#define AST_RTP_CISCO_DTMF (1 << 2)
/*! Maximum RTP-specific code */
-#define AST_RTP_MAX AST_RTP_CISCO_DTMF
+#define AST_RTP_MAX AST_RTP_CISCO_DTMF
+
+#define MAX_RTP_PT 256
struct ast_rtp_protocol {
/*! Get RTP struct, or NULL if unwilling to transfer */
@@ -60,15 +62,61 @@
AST_LIST_ENTRY(ast_rtp_protocol) list;
};
+typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
+
+/*! \brief The value of each payload format mapping: */
+struct rtpPayloadType {
+ int isAstFormat; /*!< whether the following code is an AST_FORMAT */
+ int code; /*!< Payload code for this type in this RTP stream */
+};
+
+#define FLAG_3389_WARNING (1 << 0)
+
/*!
* \brief Structure representing a RTP session.
*
* RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]"
*
*/
-struct ast_rtp;
-
-typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
+struct ast_rtp {
+ int s; /*!< File descriptor */
+ char resp; /*!< Frame subclass */
+ struct ast_frame f; /*!< The media frame */
+ unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET]; /*!< The RTP raw data */
+ unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
+ unsigned int lastts;
+ unsigned int lastdigitts;
+ unsigned int lastrxts;
+ unsigned int lastividtimestamp;
+ unsigned int lastovidtimestamp;
+ unsigned int lasteventseqn;
+ unsigned int lasteventendseqn;
+ int lasttxformat;
+ int lastrxformat;
+ int dtmfcount;
+ unsigned int dtmfduration;
+ int nat;
+ unsigned int flags;
+ struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
+ struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
+ struct timeval rxcore;
+ struct timeval txcore;
+ struct timeval dtmfmute;
+ struct ast_smoother *smoother;
+ int *ioid;
+ unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
+ unsigned short rxseqno;
+ struct sched_context *sched;
+ struct io_context *io;
+ void *data;
+ ast_rtp_callback callback;
+ struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
+ int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
+ int rtp_lookup_code_cache_code;
+ int rtp_lookup_code_cache_result;
+ struct ast_rtcp *rtcp;
+};
+
/*!
* \brief Initializate a RTP session.
Modified: team/oej/sdpcleanup/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/sdpcleanup/rtp.c?rev=25926&r1=25925&r2=25926&view=diff
==============================================================================
--- team/oej/sdpcleanup/rtp.c (original)
+++ team/oej/sdpcleanup/rtp.c Tue May 9 03:28:27 2006
@@ -73,58 +73,10 @@
static int nochecksums = 0;
#endif
-/*! \brief The value of each payload format mapping: */
-struct rtpPayloadType {
- int isAstFormat; /*!< whether the following code is an AST_FORMAT */
- int code;
-};
-
-#define MAX_RTP_PT 256
-
#define FLAG_3389_WARNING (1 << 0)
#define FLAG_NAT_ACTIVE (3 << 1)
#define FLAG_NAT_INACTIVE (0 << 1)
#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
-
-/*! \brief RTP session description */
-struct ast_rtp {
- int s;
- char resp;
- struct ast_frame f;
- unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
- unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
- unsigned int lastts;
- unsigned int lastdigitts;
- unsigned int lastrxts;
- unsigned int lastividtimestamp;
- unsigned int lastovidtimestamp;
- unsigned int lasteventseqn;
- unsigned int lasteventendseqn;
- int lasttxformat;
- int lastrxformat;
- int dtmfcount;
- unsigned int dtmfduration;
- int nat;
- unsigned int flags;
- struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
- struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
- struct timeval rxcore;
- struct timeval txcore;
- struct timeval dtmfmute;
- struct ast_smoother *smoother;
- int *ioid;
- unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
- unsigned short rxseqno;
- struct sched_context *sched;
- struct io_context *io;
- void *data;
- ast_rtp_callback callback;
- struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
- int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
- int rtp_lookup_code_cache_code;
- int rtp_lookup_code_cache_result;
- struct ast_rtcp *rtcp;
-};
/*!
* \brief Structure defining an RTCP session.
@@ -787,7 +739,7 @@
return 1;
}
-/*! \brief Make a note of a RTP paymoad type that was seen in a SDP "m=" line.
+/*! \brief Make a note of a RTP payload type that was seen in a SDP "m=" line.
* By default, use the well-known value for this type (although it may
* still be set to a different value by a subsequent "a=rtpmap:" line)
*/
@@ -796,9 +748,8 @@
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
- if (static_RTP_PT[pt].code != 0) {
+ if (static_RTP_PT[pt].code != 0)
rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
- }
}
/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
@@ -1378,7 +1329,7 @@
/* Make sure we have enough space for RTP header */
if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
- ast_log(LOG_WARNING, "RTP can only send voice\n");
+ ast_log(LOG_WARNING, "RTP can only send voice and video\n");
return -1;
}
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