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-rw-r--r--addons/chan_ooh323.c8
-rw-r--r--channels/chan_h323.c6
-rw-r--r--channels/chan_mgcp.c5
-rw-r--r--channels/chan_sip.c24
-rw-r--r--channels/chan_skinny.c5
-rw-r--r--channels/sip/include/sip.h3
-rw-r--r--configs/sip.conf.sample3
-rw-r--r--include/asterisk/frame.h4
-rw-r--r--include/asterisk/rtp_engine.h38
-rw-r--r--main/channel.c5
-rw-r--r--main/rtp_engine.c13
-rw-r--r--res/res_rtp_asterisk.c67
12 files changed, 122 insertions, 59 deletions
diff --git a/addons/chan_ooh323.c b/addons/chan_ooh323.c
index d7a4dfc88..7724c21d4 100644
--- a/addons/chan_ooh323.c
+++ b/addons/chan_ooh323.c
@@ -1206,10 +1206,12 @@ static int ooh323_indicate(struct ast_channel *ast, int condition, const void *d
ooManualRingback(callToken);
}
break;
- case AST_CONTROL_SRCUPDATE:
- ast_rtp_instance_new_source(p->rtp);
+ case AST_CONTROL_SRCUPDATE:
+ ast_rtp_instance_update_source(p->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_instance_change_source(p->rtp);
break;
-
case AST_CONTROL_CONNECTED_LINE:
if (gH323Debug)
ast_log(LOG_DEBUG, "Sending connected line info for %s (%s)\n",
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 145a3f5bb..f2a6bc511 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -914,7 +914,11 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_instance_new_source(pvt->rtp);
+ ast_rtp_instance_update_source(pvt->rtp);
+ res = 0;
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_instance_change_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_PROCEEDING:
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 5dfafb84a..16195cb2e 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1456,7 +1456,10 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_instance_new_source(sub->rtp);
+ ast_rtp_instance_update_source(sub->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_instance_change_source(sub->rtp);
break;
case AST_CONTROL_PROGRESS:
case AST_CONTROL_PROCEEDING:
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b8f93b583..ae1cbb3a2 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3903,7 +3903,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
if (dialog->rtp) { /* Audio */
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
/* Set Frame packetization */
@@ -3913,7 +3912,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
if (dialog->vrtp) { /* Video */
ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
- ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
@@ -4970,7 +4968,7 @@ static int sip_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
- ast_rtp_instance_new_source(p->rtp);
+ ast_rtp_instance_update_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE, TRUE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
@@ -5002,7 +5000,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_rtp_instance_new_source(p->rtp);
+ ast_rtp_instance_update_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -5333,11 +5331,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_instance_new_source(p->rtp);
+ ast_rtp_instance_update_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_instance_new_source(p->rtp);
+ ast_rtp_instance_update_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -5356,7 +5354,10 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
}
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_instance_new_source(p->rtp);
+ ast_rtp_instance_update_source(p->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_instance_change_source(p->rtp);
break;
case AST_CONTROL_CONNECTED_LINE:
update_connectedline(p, data, datalen);
@@ -19232,12 +19233,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
res = -1;
goto request_invite_cleanup;
}
- if (p->rtp) {
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
- }
- if (p->vrtp) {
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
- }
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
ast_debug(2, "No SDP in Invite, third party call control\n");
@@ -22595,9 +22590,6 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} else if (!strcasecmp(v->name, "buggymwi")) {
ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
- } else if (!strcasecmp(v->name, "constantssrc")) {
- ast_set_flag(&mask[1], SIP_PAGE2_CONSTANT_SSRC);
- ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
} else
res = 0;
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 136e2699c..a19fdc15f 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -4304,7 +4304,10 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_instance_new_source(sub->rtp);
+ ast_rtp_instance_update_source(sub->rtp);
+ break;
+ case AST_CONTROL_SRCCHANGE:
+ ast_rtp_instance_change_source(sub->rtp);
break;
case AST_CONTROL_CONNECTED_LINE:
update_connectedline(sub, data, datalen);
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 6ae1ce069..57fac84ca 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -308,7 +308,6 @@
#define SIP_PAGE2_Q850_REASON (1 << 4) /*!< DP: Get/send cause code via Reason header */
/* Space for addition of other realtime flags in the future */
-#define SIP_PAGE2_CONSTANT_SSRC (1 << 7) /*!< GDP: Don't change SSRC on reinvite */
#define SIP_PAGE2_SYMMETRICRTP (1 << 8) /*!< GDP: Whether symmetric RTP is enabled or not */
#define SIP_PAGE2_STATECHANGEQUEUE (1 << 9) /*!< D: Unsent state pending change exists */
@@ -346,7 +345,7 @@
SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
- SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP | SIP_PAGE2_CONSTANT_SSRC |\
+ SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
SIP_PAGE2_Q850_REASON)
/*@}*/
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index c4c4b83ea..a2d25f261 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -789,8 +789,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
-;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes
-
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
@@ -1003,7 +1001,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; timerb
; qualifyfreq
; t38pt_usertpsource
-; constantssrc
; contactpermit ; Limit what a host may register as (a neat trick
; contactdeny ; is to register at the same IP as a SIP provider,
; ; then call oneself, and get redirected to that
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index f9a9a4f0a..6f45386cd 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -85,7 +85,8 @@ struct ast_codec_pref {
* \arg \b HOLD Call is placed on hold
* \arg \b UNHOLD Call is back from hold
* \arg \b VIDUPDATE Video update requested
- * \arg \b SRCUPDATE The source of media has changed
+ * \arg \b SRCUPDATE The source of media has changed (RTP marker bit must change)
+ * \arg \b SRCCHANGE Media source has changed (RTP marker bit and SSRC must change)
* \arg \b CONNECTED_LINE Connected line has changed
* \arg \b REDIRECTING Call redirecting information has changed.
*/
@@ -323,6 +324,7 @@ enum ast_control_frame_type {
AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */
AST_CONTROL_REDIRECTING = 23, /*!< Indicate redirecting id has changed */
AST_CONTROL_T38_PARAMETERS = 24, /*! T38 state change request/notification with parameters */
+ AST_CONTROL_SRCCHANGE = 25, /*!< Media source has changed and requires a new RTP SSRC */
};
enum ast_control_t38 {
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 86f7c27e7..770f4d2f5 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -92,8 +92,6 @@ enum ast_rtp_property {
AST_RTP_PROPERTY_STUN,
/*! Enable RTCP support */
AST_RTP_PROPERTY_RTCP,
- /*! Don't force a new SSRC on new source */
- AST_RTP_PROPERTY_CONSTANT_SSRC,
/*!
* \brief Maximum number of RTP properties supported
@@ -322,10 +320,10 @@ struct ast_rtp_engine {
int (*dtmf_begin)(struct ast_rtp_instance *instance, char digit);
/*! Callback for stopping RFC2833 DTMF transmission */
int (*dtmf_end)(struct ast_rtp_instance *instance, char digit);
- /*! Callback to indicate that a new source of media has come in */
- void (*new_source)(struct ast_rtp_instance *instance);
- /*! Callback to tell new_source not to change SSRC */
- void (*constant_ssrc_set)(struct ast_rtp_instance *instance);
+ /*! Callback to indicate that we should update the marker bit */
+ void (*update_source)(struct ast_rtp_instance *instance);
+ /*! Callback to indicate that we should update the marker bit and ssrc */
+ void (*change_source)(struct ast_rtp_instance *instance);
/*! Callback for setting an extended RTP property */
int (*extended_prop_set)(struct ast_rtp_instance *instance, int property, void *value);
/*! Callback for getting an extended RTP property */
@@ -1192,22 +1190,40 @@ int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_r
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
/*!
- * \brief Indicate a new source of audio has dropped in
+ * \brief Indicate that the RTP marker bit should be set on an RTP stream
*
* \param instance Instance that the new media source is feeding into
*
* Example usage:
*
* \code
- * ast_rtp_instance_new_source(instance);
+ * ast_rtp_instance_update_source(instance);
* \endcode
*
- * This indicates that a new source of media is feeding the instance pointed to by
- * instance.
+ * This indicates that the source of media that is feeding the instance pointed to by
+ * instance has been updated and that the marker bit should be set.
*
* \since 1.8
*/
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance);
+void ast_rtp_instance_update_source(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Indicate a new source of audio has dropped in and the ssrc should change
+ *
+ * \param instance Instance that the new media source is feeding into
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_change_source(instance);
+ * \endcode
+ *
+ * This indicates that the source of media that is feeding the instance pointed to by
+ * instance has changed and that the marker bit should be set and the SSRC updated.
+ *
+ * \since 1.8
+ */
+void ast_rtp_instance_change_source(struct ast_rtp_instance *instance);
/*!
* \brief Set QoS parameters on an RTP session
diff --git a/main/channel.c b/main/channel.c
index 833de79b3..016453c24 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -2964,6 +2964,7 @@ int ast_waitfordigit_full(struct ast_channel *c, int ms, int audiofd, int cmdfd)
case AST_CONTROL_RINGING:
case AST_CONTROL_ANSWER:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case AST_CONTROL_CONNECTED_LINE:
case AST_CONTROL_REDIRECTING:
/* Unimportant */
@@ -3599,6 +3600,7 @@ static int attribute_const is_visible_indication(enum ast_control_frame_type con
case AST_CONTROL_PROCEEDING:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case AST_CONTROL_RADIO_KEY:
case AST_CONTROL_RADIO_UNKEY:
case AST_CONTROL_OPTION:
@@ -3739,6 +3741,7 @@ int ast_indicate_data(struct ast_channel *chan, int _condition,
case AST_CONTROL_PROCEEDING:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case AST_CONTROL_RADIO_KEY:
case AST_CONTROL_RADIO_UNKEY:
case AST_CONTROL_OPTION:
@@ -4474,6 +4477,7 @@ struct ast_channel *__ast_request_and_dial(const char *type, format_t format, co
case AST_CONTROL_UNHOLD:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
case AST_CONTROL_CONNECTED_LINE:
case AST_CONTROL_REDIRECTING:
case -1: /* Ignore -- just stopping indications */
@@ -5724,6 +5728,7 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
case AST_CONTROL_UNHOLD:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
ast_indicate_data(other, f->subclass.integer, f->data.ptr, f->datalen);
if (jb_in_use) {
ast_jb_empty_and_reset(c0, c1);
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 8085e9d9e..88f534614 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -728,10 +728,17 @@ enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *i
return instance->dtmf_mode;
}
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
+void ast_rtp_instance_update_source(struct ast_rtp_instance *instance)
{
- if (instance->engine->new_source) {
- instance->engine->new_source(instance);
+ if (instance->engine->update_source) {
+ instance->engine->update_source(instance);
+ }
+}
+
+void ast_rtp_instance_change_source(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->change_source) {
+ instance->engine->change_source(instance);
}
}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index d8a279199..fb060c75f 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -254,7 +254,8 @@ static int ast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *
static int ast_rtp_destroy(struct ast_rtp_instance *instance);
static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
-static void ast_rtp_new_source(struct ast_rtp_instance *instance);
+static void ast_rtp_update_source(struct ast_rtp_instance *instance);
+static void ast_rtp_change_source(struct ast_rtp_instance *instance);
static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
@@ -276,7 +277,8 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
.destroy = ast_rtp_destroy,
.dtmf_begin = ast_rtp_dtmf_begin,
.dtmf_end = ast_rtp_dtmf_end,
- .new_source = ast_rtp_new_source,
+ .update_source = ast_rtp_update_source,
+ .change_source = ast_rtp_change_source,
.write = ast_rtp_write,
.read = ast_rtp_read,
.prop_set = ast_rtp_prop_set,
@@ -655,16 +657,27 @@ static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
return 0;
}
-static void ast_rtp_new_source(struct ast_rtp_instance *instance)
+static void ast_rtp_update_source(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
/* We simply set this bit so that the next packet sent will have the marker bit turned on */
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+ ast_debug(3, "Setting the marker bit due to a source update\n");
- if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_CONSTANT_SSRC)) {
- rtp->ssrc = ast_random();
- }
+ return;
+}
+
+static void ast_rtp_change_source(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ unsigned int ssrc = ast_random();
+
+ /* We simply set this bit so that the next packet sent will have the marker bit turned on */
+ ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
+
+ ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
+ rtp->ssrc = ssrc;
return;
}
@@ -1854,6 +1867,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
unsigned int *rtpheader = (unsigned int*)(rtp->rawdata + AST_FRIENDLY_OFFSET), seqno, ssrc, timestamp;
struct ast_rtp_payload_type payload;
struct sockaddr_in remote_address = { 0, };
+ AST_LIST_HEAD_NOLOCK(, ast_frame) frames;
/* If this is actually RTCP let's hop on over and handle it */
if (rtcp) {
@@ -1951,14 +1965,27 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
timestamp = ntohl(rtpheader[1]);
ssrc = ntohl(rtpheader[2]);
- /* Force a marker bit if the SSRC changes */
- if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
- if (option_debug || rtpdebug) {
- ast_debug(1, "Forcing Marker bit, because SSRC has changed\n");
+ AST_LIST_HEAD_INIT_NOLOCK(&frames);
+ /* Force a marker bit and change SSRC if the SSRC changes */
+ if (rtp->rxssrc && rtp->rxssrc != ssrc) {
+ struct ast_frame *f, srcupdate = {
+ AST_FRAME_CONTROL,
+ .subclass.integer = AST_CONTROL_SRCCHANGE,
+ };
+
+ if (!mark) {
+ if (option_debug || rtpdebug) {
+ ast_debug(1, "Forcing Marker bit, because SSRC has changed\n");
+ }
+ mark = 1;
}
- mark = 1;
+
+ f = ast_frisolate(&srcupdate);
+ AST_LIST_INSERT_TAIL(&frames, f, frame_list);
}
+ rtp->rxssrc = ssrc;
+
/* Remove any padding bytes that may be present */
if (padding) {
res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
@@ -1986,7 +2013,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* Make sure after we potentially mucked with the header length that it is once again valid */
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d\n", res, hdrlen);
- return &ast_null_frame;
+ return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
}
rtp->rxcount++;
@@ -2029,7 +2056,11 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(remote_address.sin_addr));
}
- return f ? f : &ast_null_frame;
+ if (f) {
+ AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+ return AST_LIST_FIRST(&frames);
+ }
+ return &ast_null_frame;
}
rtp->lastrxformat = rtp->f.subclass.codec = payload.code;
@@ -2046,7 +2077,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.codec)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
- return f;
+ AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+ return AST_LIST_FIRST(&frames);
}
}
@@ -2081,7 +2113,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->f.subclass.codec = AST_FORMAT_T140;
header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
if (header_end == NULL) {
- return &ast_null_frame;
+ return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
}
header_end++;
@@ -2094,7 +2126,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
len += data[x * 4 + 3];
if (!(rtp->f.datalen - len))
- return &ast_null_frame;
+ return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
rtp->f.data.ptr += len;
rtp->f.datalen -= len;
@@ -2150,7 +2182,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->f.delivery.tv_usec = 0;
}
- return &rtp->f;
+ AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
+ return AST_LIST_FIRST(&frames);
}
static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-12 09:05:47 +0000