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-rw-r--r--CHANGES16
-rw-r--r--UPGRADE.txt7
-rw-r--r--apps/app_queue.c28
-rw-r--r--apps/confbridge/confbridge_manager.c2
-rw-r--r--channels/chan_sip.c139
-rw-r--r--channels/chan_skinny.c9
-rw-r--r--channels/sip/include/sip.h3
-rw-r--r--configs/samples/pjsip.conf.sample11
-rw-r--r--configs/samples/sip.conf.sample2
-rw-r--r--contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py1
-rw-r--r--contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py32
-rw-r--r--include/asterisk/network.h5
-rw-r--r--include/asterisk/res_pjsip.h80
-rw-r--r--main/http.c3
-rw-r--r--main/manager.c4
-rw-r--r--res/res_pjsip.c141
-rw-r--r--res/res_pjsip/config_transport.c22
-rw-r--r--res/res_pjsip/pjsip_message_ip_updater.c83
-rw-r--r--res/res_pjsip_pubsub.c46
-rw-r--r--res/res_pjsip_sdp_rtp.c16
-rw-r--r--res/res_pjsip_session.c34
-rw-r--r--res/res_rtp_asterisk.c12
22 files changed, 537 insertions, 159 deletions
diff --git a/CHANGES b/CHANGES
index 4e9df4fa5..e391d1e17 100644
--- a/CHANGES
+++ b/CHANGES
@@ -118,6 +118,22 @@ app_voicemail
* Added 'fromstring' field to the voicemail boxes. If set, it will override
the global 'fromstring' field on a per-mailbox basis.
+res_pjsip
+------------------
+ * A new transport parameter 'symmetric_transport' has been added.
+ When a request from a dynamic contact comes in on a transport with this
+ option set to 'yes', the transport name will be saved and used for
+ subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's
+ saved as a contact uri parameter named 'x-ast-txp' and will display with
+ the contact uri in CLI, AMI, and ARI output. On the outgoing request,
+ if a transport wasn't explicitly set on the endpoint AND the request URI
+ is not a hostname, the saved transport will be used and the 'x-ast-txp'
+ parameter stripped from the outgoing packet. To facilitate recreation of
+ subscriptions on asterisk restart, a new column 'contact_uri' needed to be
+ added to the ps_subcsription_persistence table. Since new columns were
+ added to both transport and subscription_persistence, an alembic upgrade
+ should be run to bring the database tables up to date.
+
res_pjsip_transport_websocket
------------------
* Removed non-secure websocket support. Firefox and Chrome have not allowed
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 2275580ca..1afacf2a4 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -27,9 +27,10 @@ From 14.3.0 to 14.4.0:
res_rtp_asterisk:
- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
- Data and Control Packets on a Single Port." So far, the only channel driver
- that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
- a PJSIP endpoint in pjsip.conf to enable the feature.
+ Data and Control Packets on a Single Port." For the PJSIP channel driver,
+ chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
+ to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
+ globally or on a per-peer basis in sip.conf.
New in 14.0.0
diff --git a/apps/app_queue.c b/apps/app_queue.c
index c0de00173..3886b7c7a 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -5948,6 +5948,7 @@ static void handle_bridge_enter(void *userdata, struct stasis_subscription *sub,
{
struct queue_stasis_data *queue_data = userdata;
struct ast_bridge_blob *enter_blob = stasis_message_data(msg);
+ SCOPED_AO2LOCK(lock, queue_data);
if (queue_data->dying) {
return;
@@ -6011,7 +6012,7 @@ static void handle_bridge_left(void *userdata, struct stasis_subscription *sub,
ast_debug(3, "Detected redirect of queue caller channel %s\n",
caller_snapshot->name);
- ast_queue_log(queue_data->queue->name, queue_data->caller_uniqueid, queue_data->member->membername,
+ ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
"COMPLETECALLER", "%ld|%ld|%d",
(long) (queue_data->starttime - queue_data->holdstart),
(long) (time(NULL) - queue_data->starttime), queue_data->caller_pos);
@@ -6047,16 +6048,17 @@ static void handle_blind_transfer(void *userdata, struct stasis_subscription *su
RAII_VAR(struct ast_channel_snapshot *, caller_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, member_snapshot, NULL, ao2_cleanup);
- if (queue_data->dying) {
- return;
- }
-
if (transfer_msg->result != AST_BRIDGE_TRANSFER_SUCCESS) {
return;
}
ao2_lock(queue_data);
+ if (queue_data->dying) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
if (ast_strlen_zero(queue_data->bridge_uniqueid) ||
strcmp(queue_data->bridge_uniqueid, transfer_msg->bridge->uniqueid)) {
ao2_unlock(queue_data);
@@ -6104,10 +6106,6 @@ static void handle_attended_transfer(void *userdata, struct stasis_subscription
RAII_VAR(struct ast_channel_snapshot *, caller_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, member_snapshot, NULL, ao2_cleanup);
- if (queue_data->dying) {
- return;
- }
-
if (atxfer_msg->result != AST_BRIDGE_TRANSFER_SUCCESS ||
atxfer_msg->dest_type == AST_ATTENDED_TRANSFER_DEST_THREEWAY) {
return;
@@ -6115,6 +6113,11 @@ static void handle_attended_transfer(void *userdata, struct stasis_subscription
ao2_lock(queue_data);
+ if (queue_data->dying) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
if (ast_strlen_zero(queue_data->bridge_uniqueid)) {
ao2_unlock(queue_data);
return;
@@ -6298,12 +6301,13 @@ static void handle_hangup(void *userdata, struct stasis_subscription *sub,
RAII_VAR(struct ast_channel *, chan, NULL, ao2_cleanup);
enum agent_complete_reason reason;
+ ao2_lock(queue_data);
+
if (queue_data->dying) {
+ ao2_unlock(queue_data);
return;
}
- ao2_lock(queue_data);
-
if (!strcmp(channel_blob->snapshot->uniqueid, queue_data->caller_uniqueid)) {
reason = CALLER;
} else if (!strcmp(channel_blob->snapshot->uniqueid, queue_data->member_uniqueid)) {
@@ -6332,7 +6336,7 @@ static void handle_hangup(void *userdata, struct stasis_subscription *sub,
ast_debug(3, "Detected hangup of queue %s channel %s\n", reason == CALLER ? "caller" : "member",
channel_blob->snapshot->name);
- ast_queue_log(queue_data->queue->name, queue_data->caller_uniqueid, queue_data->member->membername,
+ ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
reason == CALLER ? "COMPLETECALLER" : "COMPLETEAGENT", "%ld|%ld|%d",
(long) (queue_data->starttime - queue_data->holdstart),
(long) (time(NULL) - queue_data->starttime), queue_data->caller_pos);
diff --git a/apps/confbridge/confbridge_manager.c b/apps/confbridge/confbridge_manager.c
index a99362b33..e5db648da 100644
--- a/apps/confbridge/confbridge_manager.c
+++ b/apps/confbridge/confbridge_manager.c
@@ -189,7 +189,7 @@
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeTalking">
<managerEventInstance class="EVENT_FLAG_CALL">
- <synopsis>Raised when a confbridge participant unmutes.</synopsis>
+ <synopsis>Raised when a confbridge participant begins or ends talking.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d158b0dbd..f659a44a3 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1216,6 +1216,7 @@ static int process_sdp_o(const char *o, struct sip_pvt *p);
static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
static int process_sdp_a_sendonly(const char *a, int *sendonly);
static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
+static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested);
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
@@ -6011,7 +6012,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
ast_rtp_instance_set_qos(dialog->vrtp, global_tos_video, global_cos_video, "SIP VIDEO");
}
@@ -6031,14 +6032,14 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
/* Do not timeout text as its not constant*/
ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
}
ast_rtp_instance_set_timeout(dialog->rtp, dialog->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->rtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
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));
@@ -7752,6 +7753,15 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
return res;
}
+enum sip_media_fds {
+ SIP_AUDIO_RTP_FD,
+ SIP_AUDIO_RTCP_FD,
+ SIP_VIDEO_RTP_FD,
+ SIP_VIDEO_RTCP_FD,
+ SIP_TEXT_RTP_FD,
+ SIP_UDPTL_FD,
+};
+
/*!
* \internal
* \brief Create and initialize UDPTL for the specified dialog
@@ -7780,7 +7790,7 @@ static int initialize_udptl(struct sip_pvt *p)
/* T38 can be supported by this dialog, create it and set the derived properties */
if ((p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr))) {
if (p->owner) {
- ast_channel_set_fd(p->owner, 5, ast_udptl_fd(p->udptl));
+ ast_channel_set_fd(p->owner, SIP_UDPTL_FD, ast_udptl_fd(p->udptl));
}
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
@@ -8206,20 +8216,28 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
* UDPTL is created as needed in the lifetime of a dialog, its file
* descriptor is set in initialize_udptl */
if (i->rtp) {
- ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
- ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTP_FD, ast_rtp_instance_fd(i->rtp, 0));
+ if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, -1);
+ } else {
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(i->rtp, 1));
+ }
ast_rtp_instance_set_write_format(i->rtp, fmt);
ast_rtp_instance_set_read_format(i->rtp, fmt);
}
if (needvideo && i->vrtp) {
- ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
- ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTP_FD, ast_rtp_instance_fd(i->vrtp, 0));
+ if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, -1);
+ } else {
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(i->vrtp, 1));
+ }
}
if (needtext && i->trtp) {
- ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
+ ast_channel_set_fd(tmp, SIP_TEXT_RTP_FD, ast_rtp_instance_fd(i->trtp, 0));
}
if (i->udptl) {
- ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
+ ast_channel_set_fd(tmp, SIP_UDPTL_FD, ast_udptl_fd(i->udptl));
}
if (state == AST_STATE_RING) {
@@ -10074,6 +10092,42 @@ static int has_media_stream(struct sip_pvt *p, enum media_type m)
return 0;
}
+static void configure_rtcp(struct sip_pvt *p, struct ast_rtp_instance *instance, int which, int remote_rtcp_mux)
+{
+ int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+ int fd = -1;
+
+ if (local_rtcp_mux && remote_rtcp_mux) {
+ ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
+ } else {
+ ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
+ fd = ast_rtp_instance_fd(instance, 1);
+ }
+
+ if (p->owner) {
+ ast_channel_set_fd(p->owner, which, fd);
+ }
+}
+
+static void set_ice_components(struct sip_pvt *p, struct ast_rtp_instance *instance, int remote_rtcp_mux)
+{
+ struct ast_rtp_engine_ice *ice;
+ int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+
+ ice = ast_rtp_instance_get_ice(instance);
+ if (!ice) {
+ return;
+ }
+
+ if (local_rtcp_mux && remote_rtcp_mux) {
+ /* We both support RTCP mux. Only one ICE component necessary */
+ ice->change_components(instance, 1);
+ } else {
+ /* They either don't support RTCP mux or we don't know if they do yet. */
+ ice->change_components(instance, 2);
+ }
+}
+
/*! \brief Process SIP SDP offer, select formats and activate media channels
If offer is rejected, we will not change any properties of the call
Return 0 on success, a negative value on errors.
@@ -10132,6 +10186,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int secure_audio = FALSE;
int secure_video = FALSE;
+ /* RTCP Multiplexing */
+ int remote_rtcp_mux_audio = FALSE;
+ int remote_rtcp_mux_video = FALSE;
+
/* Others */
int sendonly = -1;
unsigned int numberofports;
@@ -10662,6 +10720,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
processed = TRUE;
+ } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_audio)) {
+ processed = TRUE;
}
}
/* Video specific scanning */
@@ -10683,6 +10743,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
processed = TRUE;
+ } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_video)) {
+ processed = TRUE;
}
}
/* Text (T.140) specific scanning */
@@ -10857,6 +10919,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (sa && portno > 0) {
/* Start ICE negotiation here, only when it is response, and setting that we are conrolling agent,
as we are offerer */
+ set_ice_components(p, p->rtp, remote_rtcp_mux_audio);
if (req->method == SIP_RESPONSE) {
start_ice(p->rtp, 1);
}
@@ -10870,11 +10933,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
/* Ensure RTCP is enabled since it may be inactive
if we're coming back from a T.38 session */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
- /* Ensure audio RTCP reads are enabled */
- if (p->owner) {
- ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
- }
+ configure_rtcp(p, p->rtp, SIP_AUDIO_RTCP_FD, remote_rtcp_mux_audio);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -10897,10 +10956,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Prevent audio RTCP reads */
if (p->owner) {
- ast_channel_set_fd(p->owner, 1, -1);
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
}
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
} else {
ast_rtp_instance_stop(p->rtp);
if (debug)
@@ -10911,6 +10970,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Setup video address and port */
if (p->vrtp) {
if (vsa && vportno > 0) {
+ set_ice_components(p, p->vrtp, remote_rtcp_mux_video);
start_ice(p->vrtp, (req->method != SIP_RESPONSE) ? 0 : 1);
ast_sockaddr_set_port(vsa, vportno);
ast_rtp_instance_set_remote_address(p->vrtp, vsa);
@@ -10919,6 +10979,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_sockaddr_stringify(vsa));
}
ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
+ configure_rtcp(p, p->vrtp, SIP_VIDEO_RTCP_FD, remote_rtcp_mux_video);
} else {
ast_rtp_instance_stop(p->vrtp);
if (debug)
@@ -11265,6 +11326,18 @@ static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_in
return found;
}
+static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested)
+{
+ int found = FALSE;
+
+ if (!strncasecmp(a, "rtcp-mux", 8)) {
+ *requested = TRUE;
+ found = TRUE;
+ }
+
+ return found;
+}
+
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_dtls *dtls;
@@ -13632,6 +13705,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
add_dtls_to_sdp(p->rtp, &a_audio);
}
+
+ /* If we've got rtcp-mux enabled, just unconditionally offer it in all SDPs */
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_str_append(&a_audio, 0, "a=rtcp-mux\r\n");
+ ast_str_append(&a_video, 0, "a=rtcp-mux\r\n");
+ }
}
if (add_t38) {
@@ -13999,18 +14078,18 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
if (p->rtp) {
if (t38version) {
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
if (p->owner) {
/* Prevent audio RTCP reads */
- ast_channel_set_fd(p->owner, 1, -1);
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
}
} else if (ast_sockaddr_isnull(&p->redirip)) {
/* Enable RTCP since it will be inactive if we're coming back
* with this reinvite */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
if (p->owner) {
/* Enable audio RTCP reads */
- ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(p->rtp, 1));
}
}
}
@@ -21021,6 +21100,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
ast_cli(fd, " Parkinglot : %s\n", peer->parkinglot);
ast_cli(fd, " Use Reason : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
ast_cli(fd, " Encryption : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)));
+ ast_cli(fd, " RTCP Mux : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX)));
ast_cli(fd, "\n");
peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer ptr");
} else if (peer && type == 1) { /* manager listing */
@@ -21091,6 +21171,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
astman_append(s, "SIP-Encryption: %s\r\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP) ? "Y" : "N");
+ astman_append(s, "SIP-RTCP-Mux: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX) ? "Y" : "N");
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -21719,6 +21800,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " MOH Interpret: %s\n", default_mohinterpret);
ast_cli(a->fd, " MOH Suggest: %s\n", default_mohsuggest);
ast_cli(a->fd, " Voice Mail Extension: %s\n", default_vmexten);
+ ast_cli(a->fd, " RTCP Multiplexing: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[2], SIP_PAGE3_RTCP_MUX)));
if (realtimepeers || realtimeregs) {
@@ -30787,6 +30869,9 @@ 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, "rtcp_mux")) {
+ ast_set_flag(&mask[2], SIP_PAGE3_RTCP_MUX);
+ ast_set2_flag(&flags[2], ast_true(v->value), SIP_PAGE3_RTCP_MUX);
} else
res = 0;
@@ -33418,9 +33503,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
if (p->rtp) {
/* Prevent audio RTCP reads */
- ast_channel_set_fd(chan, 1, -1);
+ ast_channel_set_fd(chan, SIP_AUDIO_RTCP_FD, -1);
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
}
} else if (!ast_sockaddr_isnull(&p->redirip)) {
memset(&p->redirip, 0, sizeof(p->redirip));
@@ -33432,9 +33517,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
if (p->vrtp) {
/* Prevent video RTCP reads */
- ast_channel_set_fd(chan, 3, -1);
+ ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, -1);
/* Silence RTCP while video RTP is inactive */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
}
} else if (!ast_sockaddr_isnull(&p->vredirip)) {
memset(&p->vredirip, 0, sizeof(p->vredirip));
@@ -33443,9 +33528,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
if (p->vrtp) {
/* Enable RTCP since it will be inactive if we're coming back
* from a reinvite */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
/* Enable video RTCP reads */
- ast_channel_set_fd(chan, 3, ast_rtp_instance_fd(p->vrtp, 1));
+ ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(p->vrtp, 1));
}
}
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index a3a2f87fb..d1c2b927a 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -7639,7 +7639,6 @@ static void *accept_thread(void *ignore)
struct sockaddr_in sin;
socklen_t sinlen;
struct skinnysession *s;
- struct protoent *p;
int arg = 1;
for (;;) {
@@ -7656,12 +7655,10 @@ static void *accept_thread(void *ignore)
continue;
}
- p = getprotobyname("tcp");
- if(p) {
- if( setsockopt(as, p->p_proto, TCP_NODELAY, (char *)&arg, sizeof(arg) ) < 0 ) {
- ast_log(LOG_WARNING, "Failed to set Skinny tcp connection to TCP_NODELAY mode: %s\n", strerror(errno));
- }
+ if (setsockopt(as, IPPROTO_TCP, TCP_NODELAY, (char *) &arg, sizeof(arg)) < 0) {
+ ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on Skinny TCP connection: %s\n", strerror(errno));
}
+
if (!(s = ast_calloc(1, sizeof(struct skinnysession)))) {
close(as);
ast_atomic_fetchadd_int(&unauth_sessions, -1);
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index e511d139b..86f8967c4 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -384,11 +384,12 @@
#define SIP_PAGE3_IGNORE_PREFCAPS (1 << 7) /*!< DP: Ignore prefcaps when setting up an outgoing call leg */
#define SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL (1 << 8) /*!< DGP: Stop telling the peer to start music on hold */
#define SIP_PAGE3_FORCE_AVP (1 << 9) /*!< DGP: Force 'RTP/AVP' for all streams, even DTLS */
+#define SIP_PAGE3_RTCP_MUX (1 << 10) /*!< DGP: Attempt to negotiate RFC 5761 RTCP multiplexing */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF | SIP_PAGE3_ICE_SUPPORT | SIP_PAGE3_IGNORE_PREFCAPS | \
- SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP)
+ SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP | SIP_PAGE3_RTCP_MUX)
#define CHECK_AUTH_BUF_INITLEN 256
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index b18fdb276..120a7ef1c 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -853,6 +853,17 @@
; this option is set to 'no' (the default) changes to the
; particular transport will be ignored. If set to 'yes',
; changes (if any) will be applied.
+;symmetric_transport=no ; When a request from a dynamic contact comes in on a
+ ; transport with this option set to 'yes', the transport
+ ; name will be saved and used for subsequent outgoing
+ ; requests like OPTIONS, NOTIFY and INVITE. It's saved
+ ; as a contact uri parameter named 'x-ast-txp' and will
+ ; display with the contact uri in CLI, AMI, and ARI
+ ; output. On the outgoing request, if a transport
+ ; wasn't explicitly set on the endpoint AND the request
+ ; URI is not a hostname, the saved transport will be
+ ; used and the 'x-ast-txp' parameter stripped from the
+ ; outgoing packet.
;==========================AOR SECTION OPTIONS=========================
;[aor]
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index 916e2d671..9b52ec06c 100644
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -1090,6 +1090,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; option may be specified at the global or peer scope.
;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
; media streams when appropriate, even if a DTLS stream is present.
+;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for
+ ; WebRTC support
; ---------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
diff --git a/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py b/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py
index 50d3ee338..8b0214a17 100644
--- a/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py
+++ b/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py
@@ -12,6 +12,7 @@ down_revision = '465e70e8c337'
from alembic import op
import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
YESNO_NAME = 'yesno_values'
YESNO_VALUES = ['yes', 'no']
diff --git a/contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py b/contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py
new file mode 100644
index 000000000..51b5066f5
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py
@@ -0,0 +1,32 @@
+"""symmetric_transport
+
+Revision ID: f638dbe2eb23
+Revises: 15db7b91a97a
+Create Date: 2017-03-09 09:38:59.513479
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'f638dbe2eb23'
+down_revision = '15db7b91a97a'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+ ############################# Enums ##############################
+
+ # yesno_values have already been created, so use postgres enum object
+ # type to get around "already created" issue - works okay with mysql
+ yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+ op.add_column('ps_transports', sa.Column('symmetric_transport', yesno_values))
+ op.add_column('ps_subscription_persistence', sa.Column('contact_uri', sa.String(256)))
+
+def downgrade():
+ op.drop_column('ps_subscription_persistence', 'contact_uri')
+ op.drop_column('ps_transports', 'symmetric_transport')
diff --git a/include/asterisk/network.h b/include/asterisk/network.h
index 3371e5895..5216f4c61 100644
--- a/include/asterisk/network.h
+++ b/include/asterisk/network.h
@@ -86,6 +86,11 @@ const char *ast_inet_ntoa(struct in_addr ia);
#endif
#define inet_ntoa __dont__use__inet_ntoa__use__ast_inet_ntoa__instead__
+#ifdef getprotobyname
+#undef getprotobyname
+#endif
+#define getprotobyname __getprotobyname_is_not_threadsafe__do_not_use__
+
/*! \brief Compares the source address and port of two sockaddr_in */
static force_inline int inaddrcmp(const struct sockaddr_in *sin1, const struct sockaddr_in *sin2)
{
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index fb0451307..c6c308bee 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -194,6 +194,8 @@ struct ast_sip_transport {
int write_timeout;
/*! Allow reload */
int allow_reload;
+ /*! Automatically send requests out the same transport requests have come in on */
+ int symmetric_transport;
};
#define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
@@ -765,6 +767,10 @@ struct ast_sip_endpoint {
unsigned int asymmetric_rtp_codec;
};
+/*! URI parameter for symmetric transport */
+#define AST_SIP_X_AST_TXP "x-ast-txp"
+#define AST_SIP_X_AST_TXP_LEN 9
+
/*!
* \brief Initialize an auth vector with the configured values.
*
@@ -1659,6 +1665,26 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
/*!
* \brief General purpose method for creating an rdata structure using specific information
+ * \since 13.15.0
+ *
+ * \param rdata[out] The rdata structure that will be populated
+ * \param packet A SIP message
+ * \param src_name The source IP address of the message
+ * \param src_port The source port of the message
+ * \param transport_type The type of transport the message was received on
+ * \param local_name The local IP address the message was received on
+ * \param local_port The local port the message was received on
+ * \param contact_uri The contact URI of the message
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ */
+int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet,
+ const char *src_name, int src_port, char *transport_type, const char *local_name,
+ int local_port, const char *contact_uri);
+
+/*!
+ * \brief General purpose method for creating an rdata structure using specific information
*
* \param rdata[out] The rdata structure that will be populated
* \param packet A SIP message
@@ -1671,8 +1697,8 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
* \retval 0 success
* \retval -1 failure
*/
-int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port, char *transport_type,
- const char *local_name, int local_port);
+int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name,
+ int src_port, char *transport_type, const char *local_name, int local_port);
/*!
* \brief General purpose method for creating a SIP request
@@ -2709,4 +2735,54 @@ void ast_sip_modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr,
void ast_sip_get_unidentified_request_thresholds(unsigned int *count, unsigned int *period,
unsigned int *prune_interval);
+/*!
+ * \brief Get the transport name from an endpoint or request uri
+ * \since 13.15.0
+ *
+ * \param endpoint
+ * \param sip_uri
+ * \param buf Buffer to receive transport name
+ * \param buf_len Buffer length
+ *
+ * \retval 0 Success
+ * \retval -1 Failure
+ *
+ * \note
+ * If endpoint->transport is not NULL, it is returned in buf.
+ * Otherwise if sip_uri has an 'x-ast-txp' parameter AND the sip_uri host is
+ * an ip4 or ip6 address, its value is returned,
+ */
+int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, char *buf, size_t buf_len);
+
+/*!
+ * \brief Sets pjsip_tpselector from an endpoint or uri
+ * \since 13.15.0
+ *
+ * \param endpoint If endpoint->transport is set, it's used
+ * \param sip_uri If sip_uri contains a x-ast-txp parameter, it's used
+ * \param selector The selector to be populated
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ */
+int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, pjsip_tpselector *selector);
+
+/*!
+ * \brief Set the transport on a dialog
+ * \since 13.15.0
+ *
+ * \param endpoint
+ * \param dlg
+ * \param selector (optional)
+ *
+ * \note
+ * This API calls ast_sip_get_transport_name(endpoint, dlg->target) and if the result is
+ * non-NULL, calls pjsip_dlg_set_transport. If 'selector' is non-NULL, it is updated with
+ * the selector used.
+ */
+int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
+ pjsip_tpselector *selector);
+
#endif /* _RES_PJSIP_H */
diff --git a/main/http.c b/main/http.c
index 0db6ee7b6..ea85a2823 100644
--- a/main/http.c
+++ b/main/http.c
@@ -1917,9 +1917,8 @@ static void *httpd_helper_thread(void *data)
* This is necessary to prevent delays (caused by buffering) as we
* write to the socket in bits and pieces.
*/
- if (setsockopt(ast_iostream_get_fd(ser->stream), IPPROTO_TCP, TCP_NODELAY, (char *) &arg, sizeof(arg) ) < 0) {
+ if (setsockopt(ast_iostream_get_fd(ser->stream), IPPROTO_TCP, TCP_NODELAY, (char *) &arg, sizeof(arg)) < 0) {
ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on HTTP connection: %s\n", strerror(errno));
- ast_log(LOG_WARNING, "Some HTTP requests may be slow to respond.\n");
}
ast_iostream_nonblock(ser->stream);
diff --git a/main/manager.c b/main/manager.c
index eae1ca52a..c1d73dce7 100644
--- a/main/manager.c
+++ b/main/manager.c
@@ -6647,8 +6647,8 @@ static void *session_do(void *data)
/* here we set TCP_NODELAY on the socket to disable Nagle's algorithm.
* This is necessary to prevent delays (caused by buffering) as we
* write to the socket in bits and pieces. */
- if (setsockopt(ast_iostream_get_fd(ser->stream), IPPROTO_TCP, TCP_NODELAY, (char *)&arg, sizeof(arg) ) < 0) {
- ast_log(LOG_WARNING, "Failed to set manager tcp connection to TCP_NODELAY mode: %s\nSome manager actions may be slow to respond.\n", strerror(errno));
+ if (setsockopt(ast_iostream_get_fd(ser->stream), IPPROTO_TCP, TCP_NODELAY, (char *) &arg, sizeof(arg)) < 0) {
+ ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on manager connection: %s\n", strerror(errno));
}
ast_iostream_nonblock(ser->stream);
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index f4df49836..962c4be4f 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -1193,6 +1193,22 @@
in-progress calls.</para>
</description>
</configOption>
+ <configOption name="symmetric_transport" default="no">
+ <synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis>
+ <description>
+ <para>When a request from a dynamic contact
+ comes in on a transport with this option set to 'yes',
+ the transport name will be saved and used for subsequent
+ outgoing requests like OPTIONS, NOTIFY and INVITE. It's
+ saved as a contact uri parameter named 'x-ast-txp' and will
+ display with the contact uri in CLI, AMI, and ARI output.
+ On the outgoing request, if a transport wasn't explicitly
+ set on the endpoint AND the request URI is not a hostname,
+ the saved transport will be used and the 'x-ast-txp'
+ parameter stripped from the outgoing packet.
+ </para>
+ </description>
+ </configOption>
</configObject>
<configObject name="contact">
<synopsis>A way of creating an aliased name to a SIP URI</synopsis>
@@ -2762,7 +2778,54 @@ pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
return ast_pjsip_endpoint;
}
-static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
+int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, char *buf, size_t buf_len)
+{
+ char *host = NULL;
+ static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
+ pjsip_param *x_transport;
+
+ if (!ast_strlen_zero(endpoint->transport)) {
+ ast_copy_string(buf, endpoint->transport, buf_len);
+ return 0;
+ }
+
+ x_transport = pjsip_param_find(&sip_uri->other_param, &x_name);
+ if (!x_transport) {
+ return -1;
+ }
+
+ /* Only use x_transport if the uri host is an ip (4 or 6) address */
+ host = ast_alloca(sip_uri->host.slen + 1);
+ ast_copy_pj_str(host, &sip_uri->host, sip_uri->host.slen + 1);
+ if (!ast_sockaddr_parse(NULL, host, PARSE_PORT_FORBID)) {
+ return -1;
+ }
+
+ ast_copy_pj_str(buf, &x_transport->value, buf_len);
+
+ return 0;
+}
+
+int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
+ pjsip_tpselector *selector)
+{
+ pjsip_sip_uri *uri;
+ pjsip_tpselector sel = { .type = PJSIP_TPSELECTOR_NONE, };
+
+ uri = pjsip_uri_get_uri(dlg->target);
+ if (!selector) {
+ selector = &sel;
+ }
+
+ ast_sip_set_tpselector_from_ep_or_uri(endpoint, uri, selector);
+ pjsip_dlg_set_transport(dlg, selector);
+
+ return 0;
+}
+
+static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user,
+ const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
{
pj_str_t tmp, local_addr;
pjsip_uri *uri;
@@ -2892,15 +2955,16 @@ int ast_sip_set_tpselector_from_transport_name(const char *transport_name, pjsip
return ast_sip_set_tpselector_from_transport(transport, selector);
}
-static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
+int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, pjsip_tpselector *selector)
{
- const char *transport_name = endpoint->transport;
+ char transport_name[128];
- if (ast_strlen_zero(transport_name)) {
+ if (ast_sip_get_transport_name(endpoint, sip_uri, transport_name, sizeof(transport_name))) {
return 0;
}
- return ast_sip_set_tpselector_from_transport_name(endpoint->transport, selector);
+ return ast_sip_set_tpselector_from_transport_name(transport_name, selector);
}
void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri)
@@ -2908,8 +2972,8 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
pjsip_sip_uri *sip_uri;
int i = 0;
pjsip_param *param;
- const pj_str_t STR_USER = { "user", 4 };
- const pj_str_t STR_PHONE = { "phone", 5 };
+ static const pj_str_t STR_USER = { "user", 4 };
+ static const pj_str_t STR_PHONE = { "phone", 5 };
if (!endpoint || !endpoint->usereqphone || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
return;
@@ -2942,7 +3006,8 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
pj_list_insert_before(&sip_uri->other_param, param);
}
-pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
+pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
+ const char *uri, const char *request_user)
{
char enclosed_uri[PJSIP_MAX_URL_SIZE];
pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
@@ -2967,12 +3032,13 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
return NULL;
}
- if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
- pjsip_dlg_terminate(dlg);
- return NULL;
- }
+ /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
+ dlg->sess_count++;
+
+ ast_sip_dlg_set_transport(endpoint, dlg, &selector);
if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
+ dlg->sess_count--;
pjsip_dlg_terminate(dlg);
return NULL;
}
@@ -3008,11 +3074,6 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target);
ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->remote.info->uri);
- /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
- dlg->sess_count++;
-
- pjsip_dlg_set_transport(dlg, &selector);
-
if (!ast_strlen_zero(outbound_proxy)) {
pjsip_route_hdr route_set, *route;
static const pj_str_t ROUTE_HNAME = { "Route", 5 };
@@ -3081,10 +3142,13 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
pjsip_transport *transport;
+ pjsip_contact_hdr *contact_hdr;
ast_assert(status != NULL);
- if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
+ contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+ if (ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(contact_hdr->uri),
+ &selector)) {
return NULL;
}
@@ -3130,8 +3194,8 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
return dlg;
}
-int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
- char *transport_type, const char *local_name, int local_port)
+int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
+ char *transport_type, const char *local_name, int local_port, const char *contact)
{
pj_str_t tmp;
@@ -3155,6 +3219,16 @@ int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_nam
return -1;
}
+ if (!ast_strlen_zero(contact)) {
+ pjsip_contact_hdr *contact_hdr;
+
+ contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+ if (contact_hdr) {
+ contact_hdr->uri = pjsip_parse_uri(rdata->tp_info.pool, (char *)contact,
+ strlen(contact), PJSIP_PARSE_URI_AS_NAMEADDR);
+ }
+ }
+
pj_strdup2(rdata->tp_info.pool, &rdata->msg_info.via->recvd_param, rdata->pkt_info.src_name);
rdata->msg_info.via->rport_param = -1;
@@ -3166,6 +3240,13 @@ int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_nam
return 0;
}
+int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
+ char *transport_type, const char *local_name, int local_port)
+{
+ return ast_sip_create_rdata_with_contact(rdata, packet, src_name, src_port, transport_type,
+ local_name, local_port, NULL);
+}
+
/* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
@@ -3247,14 +3328,6 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
pj_cstr(&remote_uri, uri);
}
- if (endpoint) {
- if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
- ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
- ast_sorcery_object_get_id(endpoint));
- return -1;
- }
- }
-
pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
if (!pool) {
@@ -3272,6 +3345,8 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
return -1;
}
+ ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(sip_uri), &selector);
+
fromuser = endpoint ? (!ast_strlen_zero(endpoint->fromuser) ? endpoint->fromuser : ast_sorcery_object_get_id(endpoint)) : NULL;
if (sip_dialog_create_from(pool, &from, fromuser,
endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
@@ -3291,6 +3366,8 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
return -1;
}
+ pjsip_tx_data_set_transport(*tdata, &selector);
+
if (endpoint && !ast_strlen_zero(endpoint->contact_user)){
pjsip_contact_hdr *contact_hdr;
pjsip_sip_uri *contact_uri;
@@ -3332,6 +3409,8 @@ int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
{
const pjsip_method *pmethod = get_pjsip_method(method);
+ ast_assert(endpoint != NULL);
+
if (!pmethod) {
ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
return -1;
@@ -3596,7 +3675,6 @@ static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
struct send_request_wrapper *req_wrapper;
pj_status_t ret_val;
pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
- pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
if (!cb && token) {
/* Silly. Without a callback we cannot do anything with token. */
@@ -3621,11 +3699,6 @@ static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
/* Add a reference to tdata. The wrapper destructor cleans it up. */
pjsip_tx_data_add_ref(tdata);
- if (endpoint) {
- sip_get_tpselector_from_endpoint(endpoint, &selector);
- pjsip_tx_data_set_transport(tdata, &selector);
- }
-
if (timeout > 0) {
pj_time_val timeout_timer_val = { timeout / 1000, timeout % 1000 };
diff --git a/res/res_pjsip/config_transport.c b/res/res_pjsip/config_transport.c
index 60b4507cd..3c41f175a 100644
--- a/res/res_pjsip/config_transport.c
+++ b/res/res_pjsip/config_transport.c
@@ -552,13 +552,20 @@ static int transport_apply(const struct ast_sorcery *sorcery, void *obj)
}
}
- if (res == PJ_SUCCESS && (transport->tos || transport->cos)) {
- pj_sock_t sock;
- pj_qos_params qos_params;
- sock = pjsip_udp_transport_get_socket(temp_state->state->transport);
- pj_sock_get_qos_params(sock, &qos_params);
- set_qos(transport, &qos_params);
- pj_sock_set_qos_params(sock, &qos_params);
+ if (res == PJ_SUCCESS) {
+ temp_state->state->transport->info = pj_pool_alloc(temp_state->state->transport->pool,
+ (AST_SIP_X_AST_TXP_LEN + strlen(transport_id) + 2));
+
+ sprintf(temp_state->state->transport->info, "%s:%s", AST_SIP_X_AST_TXP, transport_id);
+
+ if (transport->tos || transport->cos) {
+ pj_sock_t sock;
+ pj_qos_params qos_params;
+ sock = pjsip_udp_transport_get_socket(temp_state->state->transport);
+ pj_sock_get_qos_params(sock, &qos_params);
+ set_qos(transport, &qos_params);
+ pj_sock_set_qos_params(sock, &qos_params);
+ }
}
} else if (transport->type == AST_TRANSPORT_TCP) {
pjsip_tcp_transport_cfg cfg;
@@ -1375,6 +1382,7 @@ int ast_sip_initialize_sorcery_transport(void)
ast_sorcery_object_field_register(sorcery, "transport", "cos", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_transport, cos));
ast_sorcery_object_field_register(sorcery, "transport", "websocket_write_timeout", AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT_STR, OPT_INT_T, PARSE_IN_RANGE, FLDSET(struct ast_sip_transport, write_timeout), 1, INT_MAX);
ast_sorcery_object_field_register(sorcery, "transport", "allow_reload", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_transport, allow_reload));
+ ast_sorcery_object_field_register(sorcery, "transport", "symmetric_transport", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_transport, symmetric_transport));
internal_sip_register_endpoint_formatter(&endpoint_transport_formatter);
diff --git a/res/res_pjsip/pjsip_message_ip_updater.c b/res/res_pjsip/pjsip_message_ip_updater.c
index 7671ad0a7..864d898b3 100644
--- a/res/res_pjsip/pjsip_message_ip_updater.c
+++ b/res/res_pjsip/pjsip_message_ip_updater.c
@@ -28,6 +28,7 @@
#define MOD_DATA_RESTRICTIONS "restrictions"
static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata);
+static pj_bool_t multihomed_on_rx_message(pjsip_rx_data *rdata);
/*! \brief Outgoing message modification restrictions */
struct multihomed_message_restrictions {
@@ -41,6 +42,7 @@ static pjsip_module multihomed_module = {
.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 1,
.on_tx_request = multihomed_on_tx_message,
.on_tx_response = multihomed_on_tx_message,
+ .on_rx_request = multihomed_on_rx_message,
};
/*! \brief Helper function to get (or allocate if not already present) restrictions on a message */
@@ -151,6 +153,44 @@ static int multihomed_rewrite_sdp(struct pjmedia_sdp_session *sdp)
return 0;
}
+static void sanitize_tdata(pjsip_tx_data *tdata)
+{
+ static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
+ pjsip_param *x_transport;
+ pjsip_sip_uri *uri;
+ pjsip_fromto_hdr *fromto;
+ pjsip_contact_hdr *contact;
+ pjsip_hdr *hdr;
+
+ if (tdata->msg->type == PJSIP_REQUEST_MSG) {
+ uri = pjsip_uri_get_uri(tdata->msg->line.req.uri);
+ x_transport = pjsip_param_find(&uri->other_param, &x_name);
+ if (x_transport) {
+ pj_list_erase(x_transport);
+ }
+ }
+
+ for (hdr = tdata->msg->hdr.next; hdr != &tdata->msg->hdr; hdr = hdr->next) {
+ if (hdr->type == PJSIP_H_TO || hdr->type == PJSIP_H_FROM) {
+ fromto = (pjsip_fromto_hdr *) hdr;
+ uri = pjsip_uri_get_uri(fromto->uri);
+ x_transport = pjsip_param_find(&uri->other_param, &x_name);
+ if (x_transport) {
+ pj_list_erase(x_transport);
+ }
+ } else if (hdr->type == PJSIP_H_CONTACT) {
+ contact = (pjsip_contact_hdr *) hdr;
+ uri = pjsip_uri_get_uri(contact->uri);
+ x_transport = pjsip_param_find(&uri->other_param, &x_name);
+ if (x_transport) {
+ pj_list_erase(x_transport);
+ }
+ }
+ }
+
+ pjsip_tx_data_invalidate_msg(tdata);
+}
+
static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
{
struct multihomed_message_restrictions *restrictions = ast_sip_mod_data_get(tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS);
@@ -159,6 +199,8 @@ static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
pjsip_via_hdr *via;
pjsip_fromto_hdr *from;
+ sanitize_tdata(tdata);
+
/* Use the destination information to determine what local interface this message will go out on */
pjsip_tpmgr_fla2_param_default(&prm);
prm.tp_type = tdata->tp_info.transport->key.type;
@@ -273,6 +315,47 @@ static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
return PJ_SUCCESS;
}
+static pj_bool_t multihomed_on_rx_message(pjsip_rx_data *rdata)
+{
+ pjsip_contact_hdr *contact;
+ pjsip_sip_uri *uri;
+ const char *transport_id;
+ struct ast_sip_transport *transport;
+ pjsip_param *x_transport;
+
+ if (rdata->msg_info.msg->type != PJSIP_REQUEST_MSG) {
+ return PJ_FALSE;
+ }
+
+ contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+ if (!(contact && contact->uri
+ && ast_begins_with(rdata->tp_info.transport->info, AST_SIP_X_AST_TXP ":"))) {
+ return PJ_FALSE;
+ }
+
+ uri = pjsip_uri_get_uri(contact->uri);
+
+ transport_id = rdata->tp_info.transport->info + AST_SIP_X_AST_TXP_LEN + 1;
+ transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_id);
+
+ if (!(transport && transport->symmetric_transport)) {
+ return PJ_FALSE;
+ }
+
+ x_transport = PJ_POOL_ALLOC_T(rdata->tp_info.pool, pjsip_param);
+ x_transport->name = pj_strdup3(rdata->tp_info.pool, AST_SIP_X_AST_TXP);
+ x_transport->value = pj_strdup3(rdata->tp_info.pool, transport_id);
+
+ pj_list_insert_before(&uri->other_param, x_transport);
+
+ ast_debug(1, "Set transport '%s' on %.*s from %.*s:%d\n", transport_id,
+ (int)rdata->msg_info.msg->line.req.method.name.slen,
+ rdata->msg_info.msg->line.req.method.name.ptr,
+ (int)uri->host.slen, uri->host.ptr, uri->port);
+
+ return PJ_FALSE;
+}
+
void ast_res_pjsip_cleanup_message_ip_updater(void)
{
ast_sip_unregister_service(&multihomed_module);
diff --git a/res/res_pjsip_pubsub.c b/res/res_pjsip_pubsub.c
index e90502485..f0467627e 100644
--- a/res/res_pjsip_pubsub.c
+++ b/res/res_pjsip_pubsub.c
@@ -123,6 +123,9 @@
<configOption name="expires">
<synopsis>The time at which the subscription expires</synopsis>
</configOption>
+ <configOption name="contact_uri">
+ <synopsis>The Contact URI of the dialog for the subscription</synopsis>
+ </configOption>
</configObject>
<configObject name="resource_list">
<synopsis>Resource list configuration parameters.</synopsis>
@@ -376,6 +379,8 @@ struct subscription_persistence {
char *tag;
/*! When this subscription expires */
struct timeval expires;
+ /*! Contact URI */
+ char contact_uri[PJSIP_MAX_URL_SIZE];
};
/*!
@@ -591,8 +596,8 @@ static void subscription_persistence_update(struct sip_subscription_tree *sub_tr
return;
}
- ast_debug(3, "Updating persistence for '%s->%s'\n",
- ast_sorcery_object_get_id(sub_tree->endpoint), sub_tree->root->resource);
+ ast_debug(3, "Updating persistence for '%s->%s'\n", sub_tree->persistence->endpoint,
+ sub_tree->root->resource);
dlg = sub_tree->dlg;
sub_tree->persistence->cseq = dlg->local.cseq;
@@ -600,10 +605,14 @@ static void subscription_persistence_update(struct sip_subscription_tree *sub_tr
if (rdata) {
int expires;
pjsip_expires_hdr *expires_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_EXPIRES, NULL);
+ pjsip_contact_hdr *contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
expires = expires_hdr ? expires_hdr->ivalue : DEFAULT_PUBLISH_EXPIRES;
sub_tree->persistence->expires = ast_tvadd(ast_tvnow(), ast_samp2tv(expires, 1));
+ pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR, contact_hdr->uri,
+ sub_tree->persistence->contact_uri, sizeof(sub_tree->persistence->contact_uri));
+
/* When receiving a packet on an streaming transport, it's possible to receive more than one SIP
* message at a time into the rdata->pkt_info.packet buffer. However, the rdata->msg_info.msg_buf
* will always point to the proper SIP message that is to be processed. When updating subscription
@@ -1550,8 +1559,9 @@ static int subscription_persistence_recreate(void *obj, void *arg, int flags)
pj_pool_reset(pool);
rdata.tp_info.pool = pool;
- if (ast_sip_create_rdata(&rdata, persistence->packet, persistence->src_name, persistence->src_port,
- persistence->transport_key, persistence->local_name, persistence->local_port)) {
+ if (ast_sip_create_rdata_with_contact(&rdata, persistence->packet, persistence->src_name,
+ persistence->src_port, persistence->transport_key, persistence->local_name,
+ persistence->local_port, persistence->contact_uri)) {
ast_log(LOG_WARNING, "Failed recreating '%s' subscription: The message could not be parsed\n",
persistence->endpoint);
ast_sorcery_delete(ast_sip_get_sorcery(), persistence);
@@ -1703,28 +1713,6 @@ void *ast_sip_subscription_get_header(const struct ast_sip_subscription *sub, co
return pjsip_msg_find_hdr_by_name(msg, &name, NULL);
}
-/*!
- * \internal
- * \brief Wrapper for pjsip_evsub_send_request
- *
- * This function (re)sets the transport before sending to catch cases
- * where the transport might have changed.
- *
- * If pjproject gives us the ability to resend, we'll only reset the transport
- * if PJSIP_ETPNOTAVAIL is returned from send.
- *
- * \returns pj_status_t
- */
-static pj_status_t internal_pjsip_evsub_send_request(struct sip_subscription_tree *sub_tree, pjsip_tx_data *tdata)
-{
- pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
-
- ast_sip_set_tpselector_from_transport_name(sub_tree->endpoint->transport, &selector);
- pjsip_dlg_set_transport(sub_tree->dlg, &selector);
-
- return pjsip_evsub_send_request(sub_tree->evsub, tdata);
-}
-
/* XXX This function is not used. */
struct ast_sip_subscription *ast_sip_create_subscription(const struct ast_sip_subscription_handler *handler,
struct ast_sip_endpoint *endpoint, const char *resource)
@@ -1772,7 +1760,7 @@ struct ast_sip_subscription *ast_sip_create_subscription(const struct ast_sip_su
evsub = sub_tree->evsub;
if (pjsip_evsub_initiate(evsub, NULL, -1, &tdata) == PJ_SUCCESS) {
- internal_pjsip_evsub_send_request(sub_tree, tdata);
+ pjsip_evsub_send_request(sub_tree->evsub, tdata);
} else {
/* pjsip_evsub_terminate will result in pubsub_on_evsub_state,
* being called and terminating the subscription. Therefore, we don't
@@ -1869,7 +1857,7 @@ static int sip_subscription_send_request(struct sip_subscription_tree *sub_tree,
return -1;
}
- res = internal_pjsip_evsub_send_request(sub_tree, tdata);
+ res = pjsip_evsub_send_request(sub_tree->evsub, tdata);
subscription_persistence_update(sub_tree, NULL, SUBSCRIPTION_PERSISTENCE_SEND_REQUEST);
@@ -5283,6 +5271,8 @@ static int load_module(void)
persistence_tag_str2struct, persistence_tag_struct2str, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sorcery, "subscription_persistence", "expires", "",
persistence_expires_str2struct, persistence_expires_struct2str, NULL, 0, 0);
+ ast_sorcery_object_field_register(sorcery, "subscription_persistence", "contact_uri", "", OPT_CHAR_ARRAY_T, 0,
+ CHARFLDSET(struct subscription_persistence, contact_uri));
if (apply_list_configuration(sorcery)) {
ast_sip_unregister_service(&pubsub_module);
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 75dc83929..a82475774 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -196,6 +196,20 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
media_address = &temp_media_address;
+ } else {
+ struct ast_sip_transport *transport =
+ ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
+ session->endpoint->transport);
+
+ if (transport && transport->state) {
+ char hoststr[PJ_INET6_ADDRSTRLEN];
+
+ pj_sockaddr_print(&transport->state->host, hoststr, sizeof(hoststr), 0);
+ ast_debug(1, "Transport: %s bound to host: %s, using this for media.\n",
+ session->endpoint->transport, hoststr);
+ ast_sockaddr_parse(media_address, hoststr, 0);
+ }
+ ao2_cleanup(transport);
}
if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
@@ -217,7 +231,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
}
if (!strcmp(session_media->stream_type, STR_AUDIO) &&
- (session->endpoint->media.tos_audio || session->endpoint->media.cos_video)) {
+ (session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
session->endpoint->media.cos_audio, "SIP RTP Audio");
} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 609d08dc0..de073d304 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -973,32 +973,10 @@ int ast_sip_session_refresh(struct ast_sip_session *session,
return 0;
}
-/*!
- * \internal
- * \brief Wrapper for pjsip_inv_send_msg
- *
- * This function (re)sets the transport before sending to catch cases
- * where the transport might have changed.
- *
- * If pjproject gives us the ability to resend, we'll only reset the transport
- * if PJSIP_ETPNOTAVAIL is returned from send.
- *
- * \returns pj_status_t
- */
-static pj_status_t internal_pjsip_inv_send_msg(pjsip_inv_session *inv, const char *transport_name, pjsip_tx_data *tdata)
-{
- pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
-
- ast_sip_set_tpselector_from_transport_name(transport_name, &selector);
- pjsip_dlg_set_transport(inv->dlg, &selector);
-
- return pjsip_inv_send_msg(inv, tdata);
-}
-
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
handle_outgoing_response(session, tdata);
- internal_pjsip_inv_send_msg(session->inv_session, session->endpoint->transport, tdata);
+ pjsip_inv_send_msg(session->inv_session, tdata);
return;
}
@@ -1229,7 +1207,7 @@ void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip
MOD_DATA_ON_RESPONSE, on_response);
handle_outgoing_request(session, tdata);
- internal_pjsip_inv_send_msg(session->inv_session, session->endpoint->transport, tdata);
+ pjsip_inv_send_msg(session->inv_session, tdata);
return;
}
@@ -2051,7 +2029,7 @@ static pjsip_inv_session *pre_session_setup(pjsip_rx_data *rdata, const struct a
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
}
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
return NULL;
}
return inv_session;
@@ -2222,7 +2200,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
}
}
return;
@@ -2234,7 +2212,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
}
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(inv_session);
@@ -2247,7 +2225,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
}
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(inv_session);
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index a67bc8135..88201837d 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -1577,7 +1577,7 @@ static int ast_rtp_dtls_active(struct ast_rtp_instance *instance)
static void ast_rtp_dtls_stop(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
- int rtcp_dtls_unique = (rtp->dtls.ssl != rtp->rtcp->dtls.ssl);
+ SSL *ssl = rtp->dtls.ssl;
dtls_srtp_stop_timeout_timer(instance, rtp, 0);
@@ -1595,7 +1595,7 @@ static void ast_rtp_dtls_stop(struct ast_rtp_instance *instance)
if (rtp->rtcp) {
dtls_srtp_stop_timeout_timer(instance, rtp, 1);
- if (rtp->rtcp->dtls.ssl && rtcp_dtls_unique) {
+ if (rtp->rtcp->dtls.ssl && (rtp->rtcp->dtls.ssl != ssl)) {
SSL_free(rtp->rtcp->dtls.ssl);
rtp->rtcp->dtls.ssl = NULL;
ast_mutex_destroy(&rtp->rtcp->dtls.lock);
@@ -4426,7 +4426,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
return &ast_null_frame;
}
- if (!*(read_area)) {
+ if (!*read_area) {
struct sockaddr_in addr_tmp;
struct ast_sockaddr addr_v4;
@@ -4448,7 +4448,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
return &ast_null_frame;
}
- return ast_rtcp_interpret(instance, read_area, read_area_size, &addr);
+ return ast_rtcp_interpret(instance, read_area, res, &addr);
}
static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int *rtpheader, int len, int hdrlen)
@@ -4633,7 +4633,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* This could be a multiplexed RTCP packet. If so, be sure to interpret it correctly */
if (rtcp_mux(rtp, read_area)) {
- return ast_rtcp_interpret(instance, read_area, read_area_size, &addr);
+ return ast_rtcp_interpret(instance, read_area, res, &addr);
}
/* Make sure the data that was read in is actually enough to make up an RTP packet */
@@ -5040,7 +5040,9 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
return;
}
rtp->rtcp->s = -1;
+#ifdef HAVE_OPENSSL_SRTP
rtp->rtcp->dtls.timeout_timer = -1;
+#endif
rtp->rtcp->schedid = -1;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 00:57:09 +0000