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-rw-r--r--channels/chan_sip.c23
-rw-r--r--configs/sip.conf.sample3
-rw-r--r--include/asterisk/rtp_engine.h19
-rw-r--r--main/rtp_engine.c7
-rw-r--r--res/res_rtp_asterisk.c14
5 files changed, 65 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b8e53d915..9c0076554 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1503,6 +1503,7 @@ struct sip_auth {
#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) /*!< GP: Should we clean memory from peers after expiry? */
#define SIP_PAGE2_RPID_UPDATE (1 << 3)
/* 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 */
@@ -1540,7 +1541,7 @@ struct sip_auth {
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_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP | SIP_PAGE2_CONSTANT_SSRC)
/*@}*/
@@ -5192,6 +5193,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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_timeout(dialog->rtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ ast_rtp_instance_set_constantssrc(dialog->rtp);
+ }
/* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
@@ -5199,6 +5203,9 @@ 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);
+ if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ ast_rtp_instance_set_constantssrc(dialog->vrtp);
+ }
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
@@ -20437,6 +20444,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
+ ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
} else {
p->jointcapability = p->capability;
ast_debug(1, "Hm.... No sdp for the moment\n");
@@ -20487,6 +20495,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_debug(1, "No compatible codecs for this SIP call.\n");
return -1;
}
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
+ if (p->rtp) {
+ ast_rtp_instance_set_constantssrc(p->rtp);
+ }
+ if (p->vrtp) {
+ ast_rtp_instance_set_constantssrc(p->vrtp);
+ }
+ }
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
ast_debug(2, "No SDP in Invite, third party call control\n");
@@ -23854,6 +23870,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
+ } 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;
@@ -25357,6 +25376,8 @@ static int reload_config(enum channelreloadreason reason)
} else if (!strcasecmp(v->name, "disallowed_methods")) {
char *disallow = ast_strdupa(v->value);
mark_parsed_methods(&sip_cfg.disallowed_methods, disallow);
+ } else if (!strcasecmp(v->name, "constantssrc")) {
+ ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC);
}
}
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index ba95e6355..bdd356c29 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -730,6 +730,8 @@ 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
@@ -935,6 +937,7 @@ 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/rtp_engine.h b/include/asterisk/rtp_engine.h
index 5d5ae3f7b..29070d0c7 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -317,6 +317,8 @@ struct ast_rtp_engine {
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 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 */
@@ -1183,6 +1185,23 @@ 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 Mark an RTP instance not to update SSRC on a new source
+ *
+ * \param instance Instance to update
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_constantssrc(instance);
+ * \endcode
+ *
+ * This sets the indicated instance to not update the RTP SSRC when new_source
+ * is called.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance);
+/*!
* \brief Indicate a new source of audio has dropped in
*
* \param instance Instance that the new media source is feeding into
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index cf6d2c6f2..53ed892b2 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -726,6 +726,13 @@ enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *i
return instance->dtmf_mode;
}
+void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->constant_ssrc_set) {
+ instance->engine->constant_ssrc_set(instance);
+ }
+}
+
void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
{
if (instance->engine->new_source) {
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 3abb6c686..42cce3786 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -103,6 +103,7 @@ enum strict_rtp_state {
#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
#define FLAG_NEED_MARKER_BIT (1 << 3)
#define FLAG_DTMF_COMPENSATE (1 << 4)
+#define FLAG_CONSTANT_SSRC (1 << 5)
/*! \brief RTP session description */
struct ast_rtp {
@@ -253,6 +254,7 @@ 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_set_constantssrc(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);
@@ -275,6 +277,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
.dtmf_begin = ast_rtp_dtmf_begin,
.dtmf_end = ast_rtp_dtmf_end,
.new_source = ast_rtp_new_source,
+ .constant_ssrc_set = ast_rtp_set_constantssrc,
.write = ast_rtp_write,
.read = ast_rtp_read,
.prop_set = ast_rtp_prop_set,
@@ -653,6 +656,13 @@ static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
return 0;
}
+void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ ast_set_flag(rtp, FLAG_CONSTANT_SSRC);
+}
+
static void ast_rtp_new_source(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
@@ -660,6 +670,10 @@ static void ast_rtp_new_source(struct ast_rtp_instance *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_test_flag(rtp, FLAG_CONSTANT_SSRC)) {
+ rtp->ssrc = ast_random();
+ }
+
return;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-12 00:28:43 +0000