[svn-commits] mmichelson: branch group/CCSS r252132 - in /team/group/CCSS: ./ addons/ chann...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Mar 12 16:28:26 CST 2010
Author: mmichelson
Date: Fri Mar 12 16:28:20 2010
New Revision: 252132
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=252132
Log:
Resolve conflict and reset automerge.
Modified:
team/group/CCSS/ (props changed)
team/group/CCSS/addons/chan_ooh323.c
team/group/CCSS/channels/chan_dahdi.c
team/group/CCSS/channels/chan_h323.c
team/group/CCSS/channels/chan_mgcp.c
team/group/CCSS/channels/chan_sip.c
team/group/CCSS/channels/chan_skinny.c
team/group/CCSS/channels/sip/include/sip.h
team/group/CCSS/configs/sip.conf.sample
team/group/CCSS/include/asterisk/frame.h
team/group/CCSS/include/asterisk/rtp_engine.h
team/group/CCSS/main/channel.c
team/group/CCSS/main/rtp_engine.c
team/group/CCSS/res/res_rtp_asterisk.c
Propchange: team/group/CCSS/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/CCSS/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Mar 12 16:28:20 2010
@@ -1,1 +1,1 @@
-/trunk:1-252000
+/trunk:1-252127
Modified: team/group/CCSS/addons/chan_ooh323.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/addons/chan_ooh323.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/addons/chan_ooh323.c (original)
+++ team/group/CCSS/addons/chan_ooh323.c Fri Mar 12 16:28:20 2010
@@ -1206,10 +1206,12 @@
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",
Modified: team/group/CCSS/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/chan_dahdi.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/channels/chan_dahdi.c (original)
+++ team/group/CCSS/channels/chan_dahdi.c Fri Mar 12 16:28:20 2010
@@ -17235,6 +17235,8 @@
confp->mfcr2.forced_release = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "mfcr2_immediate_accept")) {
confp->mfcr2.immediate_accept = ast_true(v->value) ? 1 : 0;
+ } else if (!strcasecmp(v->name, "mfcr2_skip_category")) {
+ confp->mfcr2.skip_category_request = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "mfcr2_call_files")) {
confp->mfcr2.call_files = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "mfcr2_max_ani")) {
Modified: team/group/CCSS/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/chan_h323.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/channels/chan_h323.c (original)
+++ team/group/CCSS/channels/chan_h323.c Fri Mar 12 16:28:20 2010
@@ -914,7 +914,11 @@
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:
Modified: team/group/CCSS/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/chan_mgcp.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/channels/chan_mgcp.c (original)
+++ team/group/CCSS/channels/chan_mgcp.c Fri Mar 12 16:28:20 2010
@@ -1456,7 +1456,10 @@
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:
Modified: team/group/CCSS/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/chan_sip.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/channels/chan_sip.c (original)
+++ team/group/CCSS/channels/chan_sip.c Fri Mar 12 16:28:20 2010
@@ -4722,7 +4722,6 @@
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 */
@@ -4732,7 +4731,6 @@
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);
@@ -5811,7 +5809,7 @@
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);
}
@@ -5843,7 +5841,7 @@
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);
@@ -6174,11 +6172,11 @@
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 */
@@ -6197,7 +6195,10 @@
}
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);
@@ -20432,12 +20433,6 @@
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");
@@ -24358,9 +24353,6 @@
} 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;
Modified: team/group/CCSS/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/chan_skinny.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/channels/chan_skinny.c (original)
+++ team/group/CCSS/channels/chan_skinny.c Fri Mar 12 16:28:20 2010
@@ -4304,7 +4304,10 @@
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);
Modified: team/group/CCSS/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/sip/include/sip.h?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/channels/sip/include/sip.h (original)
+++ team/group/CCSS/channels/sip/include/sip.h Fri Mar 12 16:28:20 2010
@@ -309,7 +309,6 @@
#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 */
@@ -347,7 +346,7 @@
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)
/*@}*/
Modified: team/group/CCSS/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/configs/sip.conf.sample?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/configs/sip.conf.sample (original)
+++ team/group/CCSS/configs/sip.conf.sample Fri Mar 12 16:28:20 2010
@@ -788,8 +788,6 @@
; also contains the Asterisk version.
;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,
@@ -1003,7 +1001,6 @@
; 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
Modified: team/group/CCSS/include/asterisk/frame.h
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/include/asterisk/frame.h?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/include/asterisk/frame.h (original)
+++ team/group/CCSS/include/asterisk/frame.h Fri Mar 12 16:28:20 2010
@@ -85,7 +85,8 @@
* \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.
*/
@@ -324,6 +325,7 @@
AST_CONTROL_REDIRECTING = 23, /*!< Indicate redirecting id has changed */
AST_CONTROL_T38_PARAMETERS = 24, /*! T38 state change request/notification with parameters */
AST_CONTROL_CC = 25, /*!< Indication that Call completion service is possible */
+ AST_CONTROL_SRCCHANGE = 26, /*!< Media source has changed and requires a new RTP SSRC */
};
enum ast_control_t38 {
Modified: team/group/CCSS/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/include/asterisk/rtp_engine.h?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/include/asterisk/rtp_engine.h (original)
+++ team/group/CCSS/include/asterisk/rtp_engine.h Fri Mar 12 16:28:20 2010
@@ -93,8 +93,6 @@
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
@@ -323,10 +321,10 @@
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 */
@@ -1193,22 +1191,40 @@
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);
- * \endcode
- *
- * This indicates that a new source of media is feeding the instance pointed to by
- * instance.
- *
- * \since 1.8
- */
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance);
+ * ast_rtp_instance_update_source(instance);
+ * \endcode
+ *
+ * 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_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
Modified: team/group/CCSS/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/main/channel.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/main/channel.c (original)
+++ team/group/CCSS/main/channel.c Fri Mar 12 16:28:20 2010
@@ -2965,6 +2965,7 @@
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 */
@@ -3600,6 +3601,7 @@
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:
@@ -3741,6 +3743,7 @@
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:
@@ -4477,6 +4480,7 @@
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 AST_CONTROL_CC:
@@ -5728,6 +5732,7 @@
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);
Modified: team/group/CCSS/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/main/rtp_engine.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/main/rtp_engine.c (original)
+++ team/group/CCSS/main/rtp_engine.c Fri Mar 12 16:28:20 2010
@@ -728,10 +728,17 @@
return instance->dtmf_mode;
}
-void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
-{
- if (instance->engine->new_source) {
- instance->engine->new_source(instance);
+void ast_rtp_instance_update_source(struct ast_rtp_instance *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);
}
}
Modified: team/group/CCSS/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/res/res_rtp_asterisk.c?view=diff&rev=252132&r1=252131&r2=252132
==============================================================================
--- team/group/CCSS/res/res_rtp_asterisk.c (original)
+++ team/group/CCSS/res/res_rtp_asterisk.c Fri Mar 12 16:28:20 2010
@@ -254,7 +254,8 @@
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 @@
.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 @@
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);
-
- if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_CONSTANT_SSRC)) {
- rtp->ssrc = ast_random();
- }
+ ast_debug(3, "Setting the marker bit due to a source update\n");
+
+ 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 @@
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) {
@@ -1951,13 +1965,26 @@
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");
- }
- mark = 1;
- }
+ 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;
+ }
+
+ 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) {
@@ -1986,7 +2013,7 @@
/* 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 @@
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 @@
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 @@
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 @@
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 @@
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)
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